"The Clue" - ny amerikansk högtalare

[Micke23]

Blev lite nyfiken på högtalaren "The Clue", någon som stött på den?

http://www.audiocircle.com/index.php?topic=86983.0

http://www.sjofnhifi.com/thecluespecs_v7.pdf

http://www.audioreview.com/cat/speakers ... 90crx.aspx

http://www.stereophile.com/content/clue-sjofn-hifi

[schmutziger]

Intressant att se att guru inte är med bland produkterna på sjofn:s hemsida längre...
Undra vad som hänt?

[Max_Headroom]

The Clue verkar intressant.

[RogerGustavsson]

Intressant att se att guru inte är med bland produkterna på sjofn:s hemsida längre...
Undra vad som hänt?

På Guru's hemsida står det under dealers:

USA
Pete Basel
Basel Consultants
basconsultants (at) aol.com

[celef]

har agenturen sjösatt en kopia, det ser så ut, då skulle jag ha blivit skitsné om jag vore guru

[Laila]

Konstruerad för Sjöfn av en "James Croft" enl. sign."sumadoggie"
på AudioCircle forum . . någon som hört talas om denne J. Croft ?

[Ragnwald]

Han är oskyldig, fick idén från ovan.
http://jamescroft.org/Podcast.htm

[Laila]

Ajsan bajsan . . . religion är strängeligen förbjudet
på forumet, har jag hört . . .

[phon]

35 hög, 20 bred och 26 djup, dom har inte alla clues än.

[Laila]

Hörrö James, va e re vi/dom, har missat . . .

[celef]

35 hög, 20 bred och 26 djup, dom har inte alla clues än.

men enligt omdömmen så spelar högtalaren fint ändå, så baffelmåtten kanske är tillräckliga för vad dom vill åstadkomma, men även elementen är ju annorlunda

[Max_Headroom]

Letar man lite på Nätet så finner man att det finns en person som heter James Croft som varit insyltad i diverse ljud-projekt.

[phon]

men enligt omdömmen så spelar högtalaren fint ändå, så baffelmåtten kanske är tillräckliga för vad dom vill åstadkomma, men även elementen är ju annorlunda

Så här verkar det se ut, elementen och baffeln.

[Ragnwald]

Ser ut som Parts Express standardlådor.
Vad är det för diskantelement?

[Lust]

"The price on The Clue, if you are in the market for high-fi equipment, is only worth mentioning because it is so absurdly low compared not only to the performance you get for the price, but also because The Clue beats the pants off both monitors and floor-standers costing 5x, 10x, 20x more. No joke. Nuff said. "

[Micke23]

[quote="Ragnwald"]Ser ut som Parts Express standardlådor.

Ja, faktiskt!

http://www.parts-express.com/pe/showdet ... TR=302-712

[phon]

Vad är det för diskantelement?

Ett Vifa med en egen waveguide tror jag.

[Lust]

Vad är det för diskantelement?

Ett Vifa med en egen waveguide tror jag.

"Some basic info on The Clue is that it is essentially a single driver design with proprietary Scanspeak woofer and tweeter. The monitor is room coupled to get bass extension flat down to between 28-32HZ depending on the "stiffness" of the wall to which it's coupled and some basic room treatment. I believe that the driver takes care of frequency response up to about 4kHZ and then there is a kind of "passive crossover" (I might be flubbing that description) which lets the tweeter take over the higher frequencies and fill in off-axis response. The mids and highs are as good as anything at the show. And all this is from a speaker that Sjofn is selling for $999 for the duration of the show."

Skrevs av sumadoggie, Audiocircle i första länken.

[zapanasta]

Diskanten ser då utseendemässigt likadan ut som den som sitter i guru qm60..

[celef]

jag skulle vilja titta in under kjolen

å diskanten är tydligen en vifa

[Lust]

å diskanten är tydligen en vifa

Verkar stämma bättre än Scan-speak...

[Ton]

"Anyway, I own a pair of the Guru QM10s that Sjofn (Lars Erickson and crew) had been repping for a few years, and The Clue (in my humble opinion) is a quantum leap beyond that monitor which has had great reviews the past couple years. The first comment out of my mouth when I heard The Clue for the first time a couple days ago was simply... CRAZY"

Borde det inte gå att skaka fram ett par i Sverige?

[RogerJoensson]

"Quantum leap".

Sedan fortsätter samme Sumadoggie:

"Yah, the WAF was absolutely taken into consideration when designing these speakers. "the clue" clearly has an aesthetic which will appeal to a large cross section of audiophiles and their spouses"

-Det var ju lite kul. -Med tanke på att de är byggda av den där standardlådan från Parts Express (om det stämmer), som dom då inte har designat...

[Aerob]

Ja, den är onekligen lik PartsExpress låda..

Från PE: Dimensions: 14" H x 8" W x 10.5" D
Från Sjofn Hifi: 14" x 8" x 10.5" (http://www.sjofnhifi.com/thecluespecs_v7.pdf)

22,5 graders invinkling ska det tydligen vara också, samt 50-55cm från golvet.

[shifts]

Det där var, om jag får säga det själv, sjukt smart.

[Fjonkalicious]

Jävligt ful låda enligt mig.

[Ton]

Jävligt ful låda enligt mig.

Ja, det tycker jag oxå. Men fruarna gillar den tydligen o så låter den helt fantastiskt. Allt enligt texten ovan

[Fjonkalicious]

Jävligt ful låda enligt mig.

Ja, det tycker jag oxå. Men fruarna gillar den tydligen o så låter den helt fantastiskt. Allt enligt texten ovan

Känns tryggt att det finns några enstaka människor i världen som på ren stortåkänsla vet hur alla fruar vill ha det.

[phon]

Borde det inte gå att skaka fram ett par i Sverige?

Det är enkelt.

En standard HiFiKit-låda. En bas och en diskant. Bygg ihop. Färdig.

[paa]

Borde det inte gå att skaka fram ett par i Sverige?

Det är enkelt.

En standard HiFiKit-låda. En bas och en diskant. Bygg ihop. Färdig.

Smart, att ingen tänkt på det tidigare?

[Laila]

Borde det inte gå att skaka fram ett par i Sverige?

Det är enkelt.

En standard HiFiKit-låda. En bas och en diskant. Bygg ihop. Färdig.

Smart, att ingen tänkt på det tidigare?

Joerå . . . redan på sjuttiotalet . . .

[phon]

[celef]

men vad kan vara anledningen till skapandet av denna högtalaren, vad är det som har hänt, har sjöfn ställt krav på guru om en mer hemmaanpassad högtalare och fått nobben, sjöfn har då blivit sura och börjat gapa å skrika för att sedan kavlat upp skjortarmarna för att skapa sig något eget?

[Ragnwald]

Hm... är det möjligen den nya i6 ?

Parts Expresslådorna är väldigt välbyggda. Zaphs använder dem flitigt.

[Objektivisten]

men vad kan vara anledningen till skapandet av denna högtalaren, vad är det som har hänt, har sjöfn ställt krav på guru om en mer hemmaanpassad högtalare och fått nobben, sjöfn har då blivit sura och börjat gapa å skrika för att sedan kavlat upp skjortarmarna för att skapa sig något eget?

I ain't gotta clue.

[Jacro]

Hello Everyone,

I am the designer of 'the clue' loudspeaker.

I noticed that there appeared to be some interest and I would be glad to discuss the clue or answer any questions that anyone might have.

I alway enjoy reading your forum, albeit with the help of Google translate.

I hope that posting in English is acceptable.

Best regards,

- James

[nilsviktor]

Vi noterar att det spekuleras om vårt förhållande med Sjöfn. Det kan vara värt att notera att vi inte haft samarbete med dem sedan hösten 2008. De hade oss med på sin hemsida längre än så, men det var inget vi kunde göra något åt.

[screen]

Vi noterar att det spekuleras om vårt förhållande med Sjöfn. Det kan vara värt att notera att vi inte haft samarbete med dem sedan hösten 2008. De hade oss med på sin hemsida längre än så, men det var inget vi kunde göra något åt.

Kan det vara värt att notera varför också?

[JanBanan]

Hello Everyone,

I am the designer of 'the clue' loudspeaker.

I noticed that there appeared to be some interest and I would be glad to discuss the clue or answer any questions that anyone might have.

I alway enjoy reading your forum, albeit with the help of Google translate.

I hope that posting in English is acceptable.

Best regards,

- James

Welcome, James!

[paa]

Hi Jacro!
Please tell us about your history with loudspeakers.

[Nattlorden]

Welcome Jacro.

If you're able to manage it with through the translator - do make a footer which tells of your commercial involvement - it's a forum rule to have it like that.

[nilsviktor]

Vi noterar att det spekuleras om vårt förhållande med Sjöfn. Det kan vara värt att notera att vi inte haft samarbete med dem sedan hösten 2008. De hade oss med på sin hemsida längre än så, men det var inget vi kunde göra något åt.

Kan det vara värt att notera varför också?

Nej. Vi tyckte olika bara.

[celef]

välkommen jacro, kul att ha dig här, berätta gärna mer om högtalarna, är det verkligen parts-express-lådor?

[Jacro]

Hi Jacro!
Please tell us about your history with loudspeakers.

First, thank you for the warm welcome!

My involvement in music reproduction goes back to childhood, but professionally it started in 1973 when after studying psychoacoustics, acoustics, and power electronics in college, as my final college project I developed a new type of Regenerative Transmission Line (currently produced by Wisdom Audio) similar to what is often referred to as a Tapped Horn.

After college I developed loudspeakers for Audionics of Oregon, the importer of the British Radford electronics at the time.

In 1975 I opened a high-end audio store called Definitive Audio, in Seattle Washington, which continues today with three stores, but I don’t participate much anymore, as I stopped running the store in 1985 and continued as Chairman of the board of directors.

With funding from Definitive Audio, in 1976 I started extensive 9-year research into psychoacoustics, acoustics, electro-acoustics, and power electronics, in an effort to define the remaining issues that keep us from having the effect of a live concert in our homes.

I’m not sure I can remember all the loudspeaker systems I’ve developed since I started professionally in about 1974 but I will list a few of the products I developed that some of you may be familiar with (I’m not sure how many products that were known in the USA were also visible in Sweden).

Audionics of Oregon:
TL-30
M33

Snell: Type One (concept and proof of concept prototypes)

Fulton (FMI) J-Modular Premiere Subwoofer and Supertweeter

Speakerlab Delta-i Hybrid

Satterberg MW2 mid-woofer for the Rogers LS3/5a

Carver Corporation:
The Carver Amazing Loudspeaker
(Planar magnetic line source/Dipole bass 12”x 4)

Carver/Toshiba Surround Sound Television Systems

JBL/Philips Projection TV Sound Systems

Dipole Surround Patent licensed to Lucasfilm/THX

Dahlquist DQ40, DQ30, DQ16, DQ6

American Technology Corp: (Now LRAD Corporation)
Hypersonic Sound (Hyper directional, Ultrasonic-to-Audio Parametric loudspeakers)

LRAD Long Range Acoustic Device (beam formed 1.5 kilometer communications)

Stratified Field; constant directivity electrostatic loudspeakers

NeoPlanar; planar magnetic transducers

The Clue

I have about 30 or 40 patents in the field of audio, mostly loudspeakers, power amplification and signal processing and continue to do consulting for many in the industry such as SJÖFN HiFI, Atlantic Technology, Magnepan and Wisdom Audio.

I hope this is the kind of answer you were looking for. If you wish to have me expand on anything in particular or answer something else, let me know.

Best regards,

James Croft

[KarlXII]

Impressive resume there, James. Jeez.


Again, Welcome to faktiskt.se.

Any chance we will see these speakers over here?

[paa]

My involvement in music reproduction goes back to childhood, but professionally it started in 1973 ...

...
Snell: Type One (concept and proof of concept prototypes)

Thanks for the resume.
Did you come up with the concept of the Snell Type One?
What is your comment to that design today? Could that floor bound tweeter solve any of todays reproduction problems, or has evolution found better ways by now?

By the way, this is my attempt at full range floor support, a mini speaker built just for fun, inspired by Stig Carlssons prototypes and patents:
http://user.faktiskt.io/paa/minione_8.jpg
I also believe Stig Carlsson was somewhat inspired by the Type One at that time.

[Jacro]

Impressive resume there, James. Jeez.


Again, Welcome to faktiskt.se.

Any chance we will see these speakers over here?

Thank you. With old age comes long resumes'.

I'm not sure of the President's (Lars Erickson) plans for export. I know he is a big fan of all things Scandinavian, and I would expect that he would like to have a good reason to visit more often. So, eventually, he may introduce The Clue in Sweden or other parts of Europe. As you may know, he is the US importer for Supra Cables.

For now, we are very busy working to build enough for the US market and it appears that will keep us busy for awhile.

Cheers,

- James

[Jacro]

My involvement in music reproduction goes back to childhood, but professionally it started in 1973 ...

...
Snell: Type One (concept and proof of concept prototypes)

Thanks for the resume.
Did you come up with the concept of the Snell Type One?
What is your comment to that design today? Could that floor bound tweeter solve any of todays reproduction problems, or has evolution found better ways by now?

By the way, this is my attempt at full range floor support, a mini speaker built just for fun, inspired by Stig Carlssons prototypes and patents:
http://user.faktiskt.io/paa/minione_8.jpg
I also believe Stig Carlsson was somewhat inspired by the Type One at that time.

Thank you for the Carlsson pic! I hadn't seen that particular model.

Yes, I came up with the concept for the Snell Type One, but in late 1976, when I did a patent search, I found Snell's original patent (US 3,964,571) had somewhat anticipated the idea, even though he was producing only the Type A at the time. So, I contacted Peter Snell and presented the concept to him, along with sketches of my configuration and test data (Mine was 8" two way, Audax and Peerless, Snell's was 10" two way Audax all around). It wasn't very nice looking, and it wasn't very practical with a 60 cm 'tongue' sticking out for people to trip over, but the performance was rather special. I had adjusted my prototypes to sound almost identical to the QUAD 57's placed on the floor.

Anyhow, I was busy with other projects and didn't want to have a patent dispute, and I was working on making a better version anyhow, as the one problem with my original approach, and the production Type One, was that the image is too low... similar to sitting up in the balcony and looking down at the orchestra. But, it was remarkably uncolored and transparent.

Two or three years later (in about 1979 I believe), Snell came out with the Type One and I offered it for sale at my audio store. Unfortunately, it went out of production after just a couple years. Even so, it was the beginning of something special and I was surprised that they abandoned it so completely. But, also, Peter Snell died around that time, so I think he would have done more with it if he had lived longer.

Yes, the concepts still have much validity today and about 12 years ago, I built prototypes of some of my next generation thinking on the idea, of which I hope to get back to in time.

With all the DIY work that the groups in this forum do, I'm sure it would be a fun project for many here to try. The patent is public domain, so have fun!

All the best,
- James

[BB]

Carver Corporation:
The Carver Amazing Loudspeaker
(Planar magnetic line source/Dipole bass 12”x 4)

James Croft

Hi James,

Cool to see that you have been working with line source systems.
Maybe we can have a chat over a beer next time I go to US..?

Best,
//BB

[paa]

[Thank you for the Carlsson pic! I hadn't seen that particular model.
...

..., and the production Type One, was that the image is too low... similar to sitting up in the balcony and looking down at the orchestra. But, it was remarkably uncolored and transparent.

Yes, the concepts still have much validity today and about 12 years ago, I built prototypes of some of my next generation thinking on the idea, of which I hope to get back to in time.

With all the DIY work that the groups in this forum do, I'm sure it would be a fun project for many here to try. The patent is public domain, so have fun!

All the best,
- James

James,
With my small speaker the sound stage can at best be quite a bit higher than one would expect, but then sometimes not.
Do you see a possibility to raise the heigth of the sound stage from loudspeakers whith the drivers placed so close to the floor?
Here is an in-depth look into the design of the never finished Carlsson Three Way speaker:
http://www.faktiskt.se/modules.php?name ... c&p=502903

[Jacro]

välkommen jacro, kul att ha dig här, berätta gärna mer om högtalarna, är det verkligen parts-express-lådor?

jacro welcome, good to have you here, tell me more about the speakers, it's really parts-express-boxes?

Celef,

I’m not sure which aspects of the design you are most interested in, so I will discuss some of the design concepts and please feel free to ask more questions if I don’t provide what you were looking for.

I believe that if the science of audio is well understood by the designer, then high quality sound doesn’t have to have high cost. Ultimately, if the design is truly advanced, the “quality” of sound should be able to be made to equal the state of the art for a modest price. With effective design, only the “quantity” of sound should drive the cost upwards, as one attempts to have the system play much louder and fill larger spaces.

The President of SJÖFN HiFi, Lars Erickson and I have known each other since the late 1970s. He and I always talked about developing a new loudspeaker that would challenge the very best, but could be affordable to those that loved music but didn’t have a lot of money to spend. We had both gone different directions for the last 25 years but we linked back up a couple years ago and immediately started where we left off, both feeling like it was time to build the speaker we had discussed so many years ago.

I had been working for a few years on new room interaction and transducer technologies and had developed a few different approaches, one of which looked like it would meet the requirements for our high value/low cost concept.

I have worked with a number approaches that attempt to ignore the room, and substantially provide direct sound only. While these can be very effective, they are at this stage in their development, rather expensive, and fairly inefficient. I wanted to have a more practical product, but one that still dealt with the “room problem”. So instead of fighting the room, and attempting to ignore it, I went in the other direction and worked to embrace the room, but to decorrelate its affects in a manner that minimized room coloration and provided great leverage to the acoustic power delivered to the listener, allowing for a modest sized system to create the level of dynamics that would be satisfying.

To start with, The Clue is based on an “enhanced” full range driver concept. The 100 mm ‘woofer’ was originally conceived as a full range driver. Full range drivers being seamless, without crossovers, have a certain desirable potential (Peter Walker of Quad was working on a full range dynamic driver as one of his last designs) but, at least in conventional form, even the few that have good on-axis response also have very poor, collapsing power response in the upper octaves, tending towards a rather ‘dark’ sound.

I wanted to have the coherence of a full range device but also with optimum power response. So, we developed the ‘woofer’ to be very smooth and wideband on axis, while using the tweeter in an unconventional manner to contribute primarily increased off-axis energy as the full-range driver dispersion starts to collapse above 2 to 4 kHz. The dispersion driver comes on stronger with increased frequency, filling in more and more of the off-axis energy while they operate together with the woofer still dominating the on axis response up to about 11 kHz wherein above that frequency, the dispersion driver starts to take over both the on and off-axis energy maintaining consistent power response to beyond 30 kHz. (I hope that is a clear enough explanation).

To further enhance the seamless transition, the dust-cap of the woofer utilizes the same silk material as the tweeter diaphragm. All of this provides a very well integrated and low coloration wave front over a consistent angle of dispersion across the entire bandwidth.

The woofer is unusual in that it has both a large linear X-max and smooth, extended upper range capability. We worked hard to get as much large signal capability (quantity) as possible while maintaining a cost effective structure. The woofer is rather expensive, but could not be built for less and still meet our bandwidth and large signal design targets.

The low frequency alignment is based on a Keele Cs = 0.58, step-down tuning, as disclosed in his 1974 AES presentation, but without requiring an active equalizer.

The very narrow use model is that of the sequential engagement of boundary gain and specific left/right speaker spacing to provide mutual coupling gain at the maximum excursion frequency. This is to passively duplicate the gain of a Q of 2 high-pass equalizer down to Fb (but with much greater acoustic power to excursion ratio), and coordinate that with a golden ratio diversification of specular boundary reflections, such that the arrival at the listener is not colored by combing of the amplitude response due to correlated reflections cancelling on-axis energy at odd half wavelengths.

This diversified, half space approach achieves very satisfying sound pressure levels while maintaining useful response to below the tuning frequency without reaching non-linear excursion from about 28 Hz upward, providing approximately a 1/3 octave deeper bass extension than the active version of the Keele alignment.

The near boundary coupling has some aspects in common with the design practice of others, such as Carlsson, Audio-Pro, and Ino-Audio, but also includes novel attributes that allow us to realize our specific target functions.

The all production units are matched within less than ½ dB to The Clue reference standard pair.

These are just a few of the design elements, and they don’t really represent a complete set of attributes that determine the sound quality of The Clue, but hopefully I have provided some useful information.

In terms of the questions about the enclosure, originally we designed a different cabinet structure with optimized internal ratios, and a form fit to the boundary coupling. In keeping with our goal of maximizing sound for cost, we found that if we modified the internal cabinet for controlled wave development and applied efficient bituminous damping to remove any remaining panel resonances, that we could use a cabinet built at the same factory as the Parts Express enclosures and equal or exceed the performance of our original custom cabinet for less than half the cost.

The goal with The Clue is to design from a systems approach, wherein increased money is spent only where it actually provides a sonic improvement and cost is reduced wherever it has no impact on sound quality.

This is achieved by way of a process that I call “Design from Local and Global Variables” using “Impedance Interface Matching”, but that is a much longer story and I have probably already carried on longer than what the questioner was looking for.

I hope this provides some of desired answers. Let me know if there are any more questions.

Cheers,

- James

[Jacro]

Carver Corporation:
The Carver Amazing Loudspeaker
(Planar magnetic line source/Dipole bass 12”x 4)

James Croft

Hi James,

Cool to see that you have been working with line source systems.
Maybe we can have a chat over a beer next time I go to US..?

Best,
//BB

Hi Bo,

I very much look forward to chatting with you about line sources and planar magnetics over a beer, or two... I'll buy.

Cheers,

- James

[Jacro]

[Thank you for the Carlsson pic! I hadn't seen that particular model.
...

..., and the production Type One, was that the image is too low... similar to sitting up in the balcony and looking down at the orchestra. But, it was remarkably uncolored and transparent.

Yes, the concepts still have much validity today and about 12 years ago, I built prototypes of some of my next generation thinking on the idea, of which I hope to get back to in time.

With all the DIY work that the groups in this forum do, I'm sure it would be a fun project for many here to try. The patent is public domain, so have fun!

All the best,
- James

James,
With my small speaker the sound stage can at best be quite a bit higher than one would expect, but then sometimes not.

Do you see a possibility to raise the heigth of the sound stage from loudspeakers whith the drivers placed so close to the floor?
Here is an in-depth look into the design of the never finished Carlsson Three Way speaker:
http://www.faktiskt.se/modules.php?name ... c&p=502903

It IS possible, by a few different methods, to raise the height of the sound stage with the speaker boundary coupled close to the floor. As with many acoustical subjects it is difficult to provide a useful and thorough answer in a few paragraphs in this forum, but I'll mention a few things to hopefully at least stimulate some thinking and discussion on the matter.

The psycho-acoustics of the situation is that we receive most of our information relative to image height from frequencies above 4 kHz, at least when we keep our head perfectly vertical. in actuality, we are always tilting our heads sideways a small amount as we listen to music and this allows our ear/brain system to start to use left/right ear comparison for vertical imaging, which lowers the frequency sensitivity down below 1 kHz for vertical detection.

Even so, one approach is to use the midrange or woofer/midrange driver floor coupled, while placing the tweeter up at ear level, crossing over at about 2 to 3 kHz with precision 8th order slope crossovers to minimize lobing in the vertical polar response at and near the crossover frequency. With this approach the tweeter provides the height information and the boundary driver operates below 3 kHz where it is most important to eliminate correlated reflections (those arriving equally at both ears, such as front wall, floor and ceiling reflections). This approach has a number of things that have to be carefully optimized to work effectively, (such as perfectly matched tweeter/midrange horizontal directivity at the crossover frequency) but it can be made to work reasonably well.

Another approach is to use both the tweeter and midrange drivers boundary coupled to the floor, and use altered frequency response above 4 kHz to mimic the pinna derived frequency changes that normally tell the ear/brain system that an image is elevated. While this can work, it tends to vary in its effectiveness depending on the listener. Additionally, the ripple that one needs to invoke to get this effect is difficult to optimize for a variety of reasons, one being that one wants the ripple to have enough amplitude and bandwidth to stimulate the ear to sense elevation of image, while at the same time not to be so high in amplitude and broad band that it colors the tonal balance.

Again, I must mention that there are many subtle issues that must be dealt with to make these types of approaches work well without side effects causing audible tonal and spatial colorations.

I hope this answers your question.

Best regards,

- James

[RogerJoensson]

with the woofer still dominating the on axis response up to about 11 kHz wherein above that frequency, the dispersion driver starts to take

So widely centered around (or mainly below?) 11 KHz you have two drivers several inches apart outputting equal levels? How have you solved the vertical dispersion, without getting deep cancellations only slightly above and below height center? -Or is this a design feature?

[Jacro]

with the woofer still dominating the on axis response up to about 11 kHz wherein above that frequency, the dispersion driver starts to take

So widely centered around (or mainly below?) 11 KHz you have two drivers several inches apart outputting equal levels? How have you solved the vertical dispersion, without getting deep cancellations only slightly above and below height center? -Or is this a design feature?

Roger,

I can share part of 'what' we are doing in this regard, but 'how' we do it must remain a trade secret.

The way this approach of enhanced power response can be made to work successfully without interference is that the outputs of the two drivers are tailored in a unique manner such that the through the overlap region, the dispersion driver contributes more off-axis energy than on-axis energy, and the full-range driver contributes more on-axis energy and less off-axis energy. The amounts that each contributes to the on or off-axis energy varies with frequency, but at any given frequency, and axis, they are not providing equal energy, so the interference is minimized and good response is maintained above and below the listening axis.

We work hard to maintain smooth response at all angles as we want to have smooth response at the listening axis and also smooth response for most room reflections, to have good first arrival response and also a balanced spectrum for the delayed, room response.

Best regards,
- James

[paa]

James, would it be possible to push this concept with a full range driver and a filler tweeter to ever larger sizes, do you think?

[Piotr]

Interesting reading, welcome to faktiskt Jacro!


/Peter

[RogerJoensson]

The way this approach of enhanced power response can be made to work successfully without interference is that the outputs of the two drivers are tailored in a unique manner such that the through the overlap region, the dispersion driver contributes more off-axis energy than on-axis energy, and the full-range driver contributes more on-axis energy and less off-axis energy. The amounts that each contributes to the on or off-axis energy varies with frequency, but at any given frequency, and axis, they are not providing equal energy, so the interference is minimized and good response is maintained above and below the listening axis.

So, in short. -The smooth/tailored power response is to heal the off-axis cancellation of the direct radiated/unreflected sound (due distance between drivers)?

[Jacro]

James, would it be possible to push this concept with a full range driver and a filler tweeter to ever larger sizes, do you think?

Generally, yes, it is scalable, but to work without compromise, the basic architecture must change with larger devices, particularly if one wishes to maintain the same transition bandwidth and corner frequencies.

For what we wanted to achieve, The Clue is the best size that still allows standard driver constructions to be effective and for the upper frequency transition frequencies to be in the right place while still being able to generate adequate low frequency extension.

- James

[Jacro]

The way this approach of enhanced power response can be made to work successfully without interference is that the outputs of the two drivers are tailored in a unique manner such that the through the overlap region, the dispersion driver contributes more off-axis energy than on-axis energy, and the full-range driver contributes more on-axis energy and less off-axis energy. The amounts that each contributes to the on or off-axis energy varies with frequency, but at any given frequency, and axis, they are not providing equal energy, so the interference is minimized and good response is maintained above and below the listening axis.

So, in short. -The smooth/tailored power response is to heal the off-axis cancellation of the direct radiated/unreflected sound (due distance between drivers)?

Well, I would say it a bit differently. The smoother power response of the combined system is to optimize the direct and off-axis responses to combine the most accurate direct arrivals while also keeping the reverberant field well balanced. By having the off axis energy delivered independently from the on axis information, I attempt to have any particular angular vector from the system maintain smooth response, but at that vector, to have the energy dominated by either the dispersion driver or the full range driver, not shared, through the overlap region, so that the interaction is minimized.

As an analogy, think of the comparison being that convention speakers have a crossover between upper and lower frequencies, wherein we are attempting to have a crossover between on-axis and off-axis energy, at least over a certain band of frequencies.

I know it is probably somewhat confusing, but it is a bit difficult to explain more clearly without telling more about how we implement the effect.

I hope this helps at least make it somewhat clearer.

All the best,

- James

[m_persson79]

It would be very interesting to listen to theese speakers.

Jacro, are they by any chance available in Denmark?

[Jacro]

It would be very interesting to listen to theese speakers.

Jacro, are they by any chance available in Denmark?

Below is my answer from the previous post with a question about exporting The Clue:
-James

KarlXII wrote:
Impressive resume there, James. Jeez.


Again, Welcome to faktiskt.se.

Any chance we will see these speakers over here?


Thank you. With old age comes long resumes'.

I'm not sure of the President's (Lars Erickson) plans for export. I know he is a big fan of all things Scandinavian, and I would expect that he would like to have a good reason to visit more often. So, eventually, he may introduce The Clue in Sweden or other parts of Europe. As you may know, he is the US importer for Supra Cables.

For now, we are very busy working to build enough for the US market and it appears that will keep us busy for awhile.

Cheers,

- James

[celef]

tack james för ett utförligt svar, jag är väldigt nyfiken på spridningstekniken i diskanten, jag likt många andra sliter hårt för att få ihop denna del i våra hembyggda högtalare, flacka delningsfilter och inte för stora element samt att försöka rikta huvudloben på sådant sätt att man minskar dips i vistelsezonen i rummet är min lösning hitills. vet du om stereophile eller soundstage kommer testa 'the clue', det vore väldigt intressant att se mätningar på dom


translate by google:

thanks james for a detailed answer, I'm very curious as to spread the technology in the treble, I like many others work hard to get together this part of our home-built speakers, flat crossover and not for major elements and to try to target the main beam in such a way as to reduces the dips in the occupied zone in the room is my solution so far. do you know about Stereophile or Soundstage will test 'the clue', it would be very interesting to see measurements on them

[patjoh]

Interesting to have you here James, good reading. Checked out your store to http://www.definitive.com/ and it looks great.

Among all the brands you sell, what is your personal favourite speakers and amplifiers? (if you are not allowed to build them yourself )

[avr7000]

Very nice to have you here!
Welcome!

Stefan

[Jacro]

tack james för ett utförligt svar, jag är väldigt nyfiken på spridningstekniken i diskanten, jag likt många andra sliter hårt för att få ihop denna del i våra hembyggda högtalare, flacka delningsfilter och inte för stora element samt att försöka rikta huvudloben på sådant sätt att man minskar dips i vistelsezonen i rummet är min lösning hitills. vet du om stereophile eller soundstage kommer testa 'the clue', det vore väldigt intressant att se mätningar på dom


translate by google:

thanks james for a detailed answer, I'm very curious as to spread the technology in the treble, I like many others work hard to get together this part of our home-built speakers, flat crossover and not for major elements and to try to target the main beam in such a way as to reduces the dips in the occupied zone in the room is my solution so far. do you know about Stereophile or Soundstage will test 'the clue', it would be very interesting to see measurements on them

Celef,

If I understand the translation properly, you are saying that you attempt to have flat response through the crossover and at the same time target the main beam (forward vertical lobe) to the location of the listener.

I'm not sure what the english translation of; "AND NOT FOR MAJOR ELEMENTS" means.

Did you mean that you try to avoid using large transducers with large center-to-center spacing? If so, I understand.

There are a number of solutions to this problem, only a very few that are effective, most approaches compromise the end result.

As you know, the vertical and horizontal power response, are both critical, but each for different reasons and each having different criteria to achieve the best sonic result.

In terms of sending the clue out for measurement, we may do so later, but right now we are not concerned so much with getting more reviews, because we are getting more orders than we can currently supply.

Our target customer is the discerning music lover who may not have the money to buy expensive loudspeakers, and may not even be very interested in the technology, but is interested in the best musical experience.

Normally, I would not be talking so much about the technical aspects of our design, as the customer that is most interested in good music reproduction isn't always interested in the technical details as much they are interested in listening.

I talked more about the technical aspects of the clue in my posts in this forum, because I know your group understands the technical concepts and likes to know more about the acoustical design elements involved (as well as good sound quality, of course!) so I answered your questions from more of a technical viewpoint.

Best regards,

- James

[paa]

James, I'll try to make an alternative translation of parts of that text, originally written by celef, where Google Translations was a bit off:

I'm very curious about the technique with the tweeter/treble dispersion, I like many others fight hard to get this thing together in our diy speakers, low order filters and not too large drivers, and trying to aim the main lobe to minimize dips in the place we dwell, is my solution so far.

[Jacro]

James, I'll try to make an alternative translation of parts of that text, originally written by celef, where Google Translations was a bit off:

I'm very curious about the technique with the tweeter/treble dispersion, I like many others fight hard to get this thing together in our diy speakers, low order filters and not too large drivers, and trying to aim the main lobe to minimize dips in the place we dwell, is my solution so far.

Yes, that is easier for me to understand. Thank you.

In general, low order filters are better if the directivity of the drivers isn't well matched at the crossover frequency.

Low order slopes tend to help make the directivity more consistent through the crossover transition frequencies, but at a cost of worse vertical lobing, so one has to steer the axis of the lobing in a manner that creates the best result.

If the directivity of the drivers is well matched, then one can use higher order slopes to minimize interference and vertical lobing, but this also requires more precision.

That is an over simplification, as there are many factors, but generally, both approaches can effective, if the variables are well understood.

Thanks again for the translation help.

- James

[Jacro]

Very nice to have you here!
Welcome!

Stefan

Stefan,

Thanks for the warm welcome.

You guys have a great forum.

- James

[celef]

translate by google

thanks for the post james! concerning the low frequency alignment and boundary gain, can you elaborate on the mutual coupling, I do not think I've read about the effect before.

and thanks for the translation paa

[Micke23]

Vilken överraskning att konstruktören plötsligt hörde av sig i tråden. Faktiskt.se har bra genomslag, det måste man säga.

[Glebster]

Audionics of Oregon:
TL-30
M33

Snell: Type One (concept and proof of concept prototypes)

Fulton (FMI) J-Modular Premiere Subwoofer and Supertweeter

Speakerlab Delta-i Hybrid

Satterberg MW2 mid-woofer for the Rogers LS3/5a

Carver Corporation:
The Carver Amazing Loudspeaker
(Planar magnetic line source/Dipole bass 12”x 4)

Carver/Toshiba Surround Sound Television Systems

JBL/Philips Projection TV Sound Systems

Dipole Surround Patent licensed to Lucasfilm/THX

Dahlquist DQ40, DQ30, DQ16, DQ6

American Technology Corp: (Now LRAD Corporation)
Hypersonic Sound (Hyper directional, Ultrasonic-to-Audio Parametric loudspeakers)

LRAD Long Range Acoustic Device (beam formed 1.5 kilometer communications)

Stratified Field; constant directivity electrostatic loudspeakers

NeoPlanar; planar magnetic transducers

The Clue

I have about 30 or 40 patents in the field of audio, mostly loudspeakers, power amplification and signal processing and continue to do consulting for many in the industry such as SJÖFN HiFI, Atlantic Technology, Magnepan and Wisdom Audio.

Respect, welcome to faktiskt.se!

[Jacro]

Audionics of Oregon:
TL-30
M33

Snell: Type One (concept and proof of concept prototypes)

Fulton (FMI) J-Modular Premiere Subwoofer and Supertweeter

Speakerlab Delta-i Hybrid

Satterberg MW2 mid-woofer for the Rogers LS3/5a

Carver Corporation:
The Carver Amazing Loudspeaker
(Planar magnetic line source/Dipole bass 12”x 4)

Carver/Toshiba Surround Sound Television Systems

JBL/Philips Projection TV Sound Systems

Dipole Surround Patent licensed to Lucasfilm/THX

Dahlquist DQ40, DQ30, DQ16, DQ6

American Technology Corp: (Now LRAD Corporation)
Hypersonic Sound (Hyper directional, Ultrasonic-to-Audio Parametric loudspeakers)

LRAD Long Range Acoustic Device (beam formed 1.5 kilometer communications)

Stratified Field; constant directivity electrostatic loudspeakers

NeoPlanar; planar magnetic transducers

The Clue

I have about 30 or 40 patents in the field of audio, mostly loudspeakers, power amplification and signal processing and continue to do consulting for many in the industry such as SJÖFN HiFI, Atlantic Technology, Magnepan and Wisdom Audio.

Respect, welcome to faktiskt.se!

Thank you Glebster, it's good to be here.
- James

[Jacro]

translate by google

thanks for the post james! concerning the low frequency alignment and boundary gain, can you elaborate on the mutual coupling, I do not think I've read about the effect before.

and thanks for the translation paa

Celef,

Mutual coupling is the effect the diaphragm of one transducer has upon another transducer diaphragm when they are closely spaced with respect to the wavelength they are radiating. There are many variables that impact the interaction, such as baffle size and nearby boundaries (front wall, sidewalls, and floor) so the information as put forth here is a simplified representation of the effect, but hopefully, useful in providing an understanding of the concept.

In general, when two transducers are approximately ½ wavelength apart, the mutual coupling starts to cause an increase in radiation resistance, which in turn causes an increased gain in output that reaches a maximum of about +3 dB at the frequency that corresponds to approximately 1/3 wavelength. This is usually considered for two drivers mounted close together in a common enclosure, but relative to the optimization of a bass alignment, we are observing the effect, in this discussion, for two separate loudspeakers and looking at the center-to-center spacing of the left loudspeaker’s woofer in relation to the right loudspeaker’s woofer. We will make the assumption, that for most program material, frequencies below 75 Hz are substantially monophonic and correlated.

As an example, two loudspeakers that are spaced apart, such that the center-to-center distance of their woofers is 2.3M, would have the effect of maximum mutual coupling gain at the frequency corresponding to 1/3 wL, which is 50 Hz (2.3 x 3 = 6.9 M = wL @ 50Hz). Gain would begin to appear at approximately ½ wL, (75Hz) and increase below that frequency until it reached maximum coupling gain at 50 Hz.
Again, if one makes the simplification of operating on an infinite plane/half-space, then one can change the spacing of the two woofers, and set the ½ wL corner frequency and 1/3 wL Maximum Gain frequency to optimize the output for a given loudspeaker and room conditions.

Keep in mind, that in the real world, actual sound rooms are not simple infinite plane/half-space environments and there are many factors that add complexity to the simplified illustration I am providing. One obvious issue is, that as one moves the loudspeakers closer together to raise the frequencies of mutual coupling gain, one is most likely also moving the them farther from the sidewalls and lowering the corner frequency of the boundary gain of the sidewall coupling.

This can be used as a method of maintaining consistent gain as one optimizes loudspeaker spacing for imaging, or it may used to shift the low frequency gain in a favorable manner for better low frequency response.

I hope this is useful.

If any of you are interested in studying subject further, here are two papers from the Journal of the Acoustical Society:

R. L. Pritchard, "Mutual Acoustic Impedance Between Radiators in an Infinite Rigid Plane",
32, 730 (1960).

S. J. Klapman, "Interaction Impedance of a System of Circular Pistons",
11, 289 (1939).

Also, if one wishes to read a more accessible (less mathematical) paper on the subject, try:
“Design Parameters of a Dual Woofer Loudspeaker System” by Edward M. Long, Journal of the Audio Engineering Society, October 1969, v.17, No. 5.

All the best,

- James

[paa]

James,
How would you comment on the horn community, with their compression drivers and horns that are designed more for low coloration and/or controlled directivity, than for maximum sensitivity and output. E.g. LeCleach, Kugelwellen, Tractix and Gedlees waveguides?
I believe there is quite a following in the US west coast and thus also Oregon for such things?
Do you see any advantages with any or all of those, compared to normal hifi-drivers for hifi and home theatre?

[Jacro]

James,
How would you comment on the horn community, with their compression drivers and horns that are designed more for low coloration and/or controlled directivity, than for maximum sensitivity and output. E.g. LeCleach, Kugelwellen, Tractix and Gedlees waveguides?
I believe there is quite a following in the US west coast and thus also Oregon for such things?
Do you see any advantages with any or all of those, compared to normal hifi-drivers for hifi and home theatre?

paa,

I believe this is some of most important work being done right now. The difference between a diffraction "horn" for maximum output, and a "waveguide" applied for the purpose of directivity control and optimized for minimal 'high order modes' is a significant distinction that has taken the high end consumer audio industry too long to recognize. Horns had developed such bad reputation relative to sound quality, that when the modern waveguide began to show promise, the audiophile community continued to reject them, associating them with the horns of the old days.

This area of development has not yet matured to reach its full potential, but as a tool for advancing the state of the art, it is one of the most promising.

LeCleach and Geddes have set good starting points in the formulations for the basic waveguide, but there are next generation architectures that are being developed to address the remaining device and systems issues. There continue to be some problems with many of the axi-symmetric devices and they need to evolve into more appropriate forms to maximize their potential, particularly in the larger devices.

With one of the key attributes of high quality loudspeaker development being that of directivity control, an effectively designed modern waveguide is one of the best tools to reach the most desirable target functions.

- James

[Nattlorden]

I've begun to wonder if it isn't just the waveguide but the entire speaker that should be machined into some bumpy blob thing to achieve desired radiation patterns...

[Naqref]

I've begun to wonder if it isn't just the waveguide but the entire speaker that should be machined into some bumpy blob thing to achieve desired radiation patterns...

Like Genelec?

[celef]

...

...

thank you for the post, i need to re-read it a few times. i have always thought about the loudspeaker as a mono-source, then interacting with the room or parts of the room, the interaction in a stereo setup is all new to me, i sure would have liked to discuss this in swedish

[paa]

LeCleach and Geddes have set good starting points in the formulations for the basic waveguide, but there are next generation architectures that are being developed to address the remaining device and systems issues. - James

Could you point to one or two from that next generation?
And would you consider the DXT lens one of those, or how would you categorize that design?
Have you tried or seen other ways to fight HOM than the foam plug that Geddes has patented?

[Nattlorden]

I've begun to wonder if it isn't just the waveguide but the entire speaker that should be machined into some bumpy blob thing to achieve desired radiation patterns...

Like Genelec?

If that would turn out to be the optimal shape, I'm more thinking about local protubations to incur beneficial difraction where needed, even if the result might look like

[Jacro]

...

...

thank you for the post, i need to re-read it a few times. i have always thought about the loudspeaker as a mono-source, then interacting with the room or parts of the room, the interaction in a stereo setup is all new to me, i sure would have liked to discuss this in swedish

Celef,

I'm sorry that I can't deliver the information in Swedish. Maybe someone in the forum can help us with translations.

I think it is important in two-channel loudspeaker design to always consider both loudspeakers to be an integrated system.

In fact, one must consider the pair of loudspeakers, the environment/room, and the listener all as an integrated, inter-dependent system. The loudspeakers are coupled locally (mutual coupling) and also coupled globally (listener and boundaries).

(In a perfect world, the program source and method of recording would be included, but not in today's discussion.)

Mono-source evaluation of a loudspeaker is certainly useful, but it is only the starting point, not a complete characterization.

There are more issues than I can cover in this post, but, as one example, even the interference and resulting frequency response ripple due to cross-talk of the two channels at the listener's head must be considered when balancing the loudspeaker 'system', and this can only be observed when both loudspeakers operating at the same time, and interaction at the listener's body, is considered.

With two-channel reproduction, there is far greater complexity of all the interactive effects, but also more possibilities to improve, not only the spatial aspects, but also the tonal effects.

Then, of course, there is another set of issues relative to loudspeaker design for more than two channels...

Cheers,

- James

[Jacro]

Welcome Jacro.

If you're able to manage it with through the translator - do make a footer which tells of your commercial involvement - it's a forum rule to have it like that.

Night Lord (Nattlorden)

Thank you for letting me know about the forum protocol.

I believe I have installed the footer information properly.

- James

[Glebster]

I believe I have installed the footer information properly.

Indeed you have!

[Laila]

Nu när vi har(ytterliggare) en bevisligen professionell
"högtalardesigner" på forumet . . . finns det verkligen
inga fler frågor att ställa till honom . . själv så diys:ar
jag inga egna högisar, men jag gissar/vet ju att flera
av medlemmarna bygger som bara attan. Om jag vore
som ni(diys:are) så skulle jag passa på innan han tröttnar
. . .(pga. ignorans)

[paa]

James, did you see my questions in this post:
http://www.faktiskt.se/modules.php?name ... 66#1111766
/paa

[Jacro]

LeCleach and Geddes have set good starting points in the formulations for the basic waveguide, but there are next generation architectures that are being developed to address the remaining device and systems issues. - James

Could you point to one or two from that next generation?
And would you consider the DXT lens one of those, or how would you categorize that design?
Have you tried or seen other ways to fight HOM than the foam plug that Geddes has patented?

Sorry for the delay in answering... I've been busy this week.

As you know, to reduce HOMs one can either;
1) Design the waveguide in a manner that minimizes the generation of high order modes,
or,
2) Apply an absorption means to reduce HOMs that are generated by the waveguide,

or both, of course.

In the first case, if a waveguide is very carefully designed to minimize high order modes then the addition of the absorption foam does not provide a significant audible improvement.

Alternatively, when designing a waveguide, if one can gain an advantage of some desirable parameter (such as improved directivity control) by allowing high order modes to increase, then adding the absorption material can make a significant sonic improvement.

This second approach can be used to improve the DXT type structure, which is very good at directivity control, but due to its directivity control being created by using diffraction steps, it can generate more high order modes than some of the other, low diffraction types, such as Oblate Spheroid, LeCleach, or Iwata types.

A good source for DIY waveguides that are designed for low HOMs (or low Bessel modes) is Auto-Tech in Poland.

http://www.autotech.pl/pdf/audio_en.pdf
http://www.diysoundgroup.com/WaveGuides.htm

I have been working on new waveguide topologies that are based on an elliptical variation of the Oblate Spheroid.

Here is a similar approach, which is a “Quasi-Super Elliptical Oblate Spheroid”.
http://www.diysoundgroup.com/images/lar ... 0-N3_1.jpg
It is also available to DIY builders.
This device has two significant advantages over the standard Geddes type axi-symmetric Oblate Spheroid. The waveguide mouth termination is asymmetrical, so it does not exhibit the problems with on-axis high frequency dips in on axis amplitude response. It also allows closer center-to-center spacing with the woofer/mid, so vertical nulls in the response can be separated farther apart and allow a larger usable vertical listening window.

This is a good improvement over the axi-symmetric approach, but doesn’t maintain and ideal Oblate Spheroid form. While it is possibly the best overall design of the current state of the art waveguides, this one is still somewhat compromised compared to an ideal for, so that it can mount flush on a flat frontal cabinet surface.

The waveguides I have been working on follow a more precise rendition of an oblate spheroid formulation converted to elliptical form, which must be shaped in a manner that doesn’t conveniently mount to a flat surface, but offers even a further reduction in modal distortions.
(It is somewhat similar in appearance to the Iwata structures).

In terms of your question about alternatives to foam as a method of reducing HOMs, I am currently exploring different materials than the standard acoustic foam insert, as per Geddes. Ideally, to reduce HOMs more than the fundamental signal output, one would want to have a angular absorption greater than axial absorption, so I am exploring new materials that can have absorption calibrated in an angle dependent manner.

Some of the first work on minimizing this type of horn/waveguide distortion by using foam was pursued by Milton Putnam in 1980, as described in his US patent 4,381,831. In this case, the absorption foam was used mainly at the mouth of the horn and on any diffractive edges within the horn.

I can speak more to this topic if there is specific interest.

Warm regards,

- James

[Jacro]

Interesting to have you here James, good reading. Checked out your store to http://www.definitive.com/ and it looks great.

Among all the brands you sell, what is your personal favourite speakers and amplifiers? (if you are not allowed to build them yourself )

Hi patjoh,

I didn't see your post until just now.

Two issues regarding your question;

First, since I am working mostly outside of Definitive Audio, focusing on research and development, I am no longer involved with many of the daily decisions, including which products are chosen to offer for sale.

Secondly, I'm sorry, but I prefer not to recommend or critique products, made by others, that are currently in production. I hope you understand.

I am more than happy to discuss classic products that are no longer being offered for sale.

Also, I am glad to discuss any design principles, psychoacoustics, acoustics, systems, or audio concepts in general and will gladly answer any design related questions anyone in the forum may wish to ask.

Best regards,

- James

[paa]

James, would you think the HOM reduction foam also works as an acoustic lens that further smothes the dispersion whithin the desired coverage angle?

[Jacro]

James, would you think the HOM reduction foam also works as an acoustic lens that further smothes the dispersion whithin the desired coverage angle?

Interesting idea.

The current goal of applying foam to a waveguide to reduce HOMs is to be able to maintain consistent absorption such that directivity is not changed from the ideal directivity of the waveguide. Correctly applied, directivity does not change when the foam is applied.

Attempting to use the foam to change directivity does add a degree of complexity that may be difficult to fully characterize. As one changes absorption rates, it must be achieved in a manner that the absorption discontinuities don't increase HOMs, simulating diffractive effects.

That said, the concept of using the foam to alter directivity while reducing HOMs may be a very good idea. The ability to provide an addition variable to control directivity in waveguides may allow one to relax the directivity capability of a waveguide in order to achieve an improvement of some other parameter (such as efficiency, or bandwidth, etc.), and then re-optimize the directivity with a variable density foam insert.

Definitely worth exploring.

- James

[celef]

...

...

thank you for the post, i need to re-read it a few times. i have always thought about the loudspeaker as a mono-source, then interacting with the room or parts of the room, the interaction in a stereo setup is all new to me, i sure would have liked to discuss this in swedish

Celef,

I'm sorry that I can't deliver the information in Swedish. Maybe someone in the forum can help us with translations.

I think it is important in two-channel loudspeaker design to always consider both loudspeakers to be an integrated system.

In fact, one must consider the pair of loudspeakers, the environment/room, and the listener all as an integrated, inter-dependent system. The loudspeakers are coupled locally (mutual coupling) and also coupled globally (listener and boundaries).

(In a perfect world, the program source and method of recording would be included, but not in today's discussion.)

Mono-source evaluation of a loudspeaker is certainly useful, but it is only the starting point, not a complete characterization.

There are more issues than I can cover in this post, but, as one example, even the interference and resulting frequency response ripple due to cross-talk of the two channels at the listener's head must be considered when balancing the loudspeaker 'system', and this can only be observed when both loudspeakers operating at the same time, and interaction at the listener's body, is considered.

With two-channel reproduction, there is far greater complexity of all the interactive effects, but also more possibilities to improve, not only the spatial aspects, but also the tonal effects.

Then, of course, there is another set of issues relative to loudspeaker design for more than two channels...

Cheers,

- James

how is this done, if i look at it at different angles i get different results for each angle!? how important is this "compensation" for the average listener? do we all have different sensitivities for errors in this regards?

[Jacro]

...

...

thank you for the post, i need to re-read it a few times. i have always thought about the loudspeaker as a mono-source, then interacting with the room or parts of the room, the interaction in a stereo setup is all new to me, i sure would have liked to discuss this in swedish

Celef,

I'm sorry that I can't deliver the information in Swedish. Maybe someone in the forum can help us with translations.

I think it is important in two-channel loudspeaker design to always consider both loudspeakers to be an integrated system.

In fact, one must consider the pair of loudspeakers, the environment/room, and the listener all as an integrated, inter-dependent system. The loudspeakers are coupled locally (mutual coupling) and also coupled globally (listener and boundaries).

(In a perfect world, the program source and method of recording would be included, but not in today's discussion.)

Mono-source evaluation of a loudspeaker is certainly useful, but it is only the starting point, not a complete characterization.

There are more issues than I can cover in this post, but, as one example, even the interference and resulting frequency response ripple due to cross-talk of the two channels at the listener's head must be considered when balancing the loudspeaker 'system', and this can only be observed when both loudspeakers operating at the same time, and interaction at the listener's body, is considered.

With two-channel reproduction, there is far greater complexity of all the interactive effects, but also more possibilities to improve, not only the spatial aspects, but also the tonal effects.

Then, of course, there is another set of issues relative to loudspeaker design for more than two channels...

Cheers,

- James

how is this done, if i look at it at different angles i get different results for each angle!? how important is this "compensation" for the average listener? do we all have different sensitivities for errors in this regards?

Celef,

I've put your questions below with answers after...


HOW IS THIS DONE?

There are a number of techniques.

One way is to simply calculate the differential in arrival distance/timing between the left and right channel to one ear. The half-wavelength (and each odd half wavelength) frequency the length differential corresponds to determines the cancellation or dip frequencies from which one can plot the ripple in the response.

Another way is to measure the differential error with a dummy head and in-ear microphones.

Another step is to compare the sound with and without a crosstalk cancelation device, using the crosstalk-cancelled mode as a reference for tonal balance.

Upon establishing the error, it is not necessary to correct all of the resulting measurable ripple. Instead, an gentle spectral correction, on a ½ to 1/3-octave basis, tends to be adequate for this type of error in that the first octave of error has a dip and peak that are a half octave apart. Above the first octave of onset, the comb filtering of the response is closely spaced and has high enough density that it doesn’t require further correction beyond the first octave, or so, of error.

The correction is a delicate one, in that if implemented inappropriately, it can create coloration in the mid-band.

As usual, this is an over simplification, and more is required to deal with the issue effectively, but hopefully this gives you the general idea.



IF I LOOK AT IT AT DIFFERENT ANGLES I GET DIFFERENT RESULTS FOR EACH ANGLE?

Yes, you are correct. The effect is different at each different listening angle.

That is one of the reasons that one must define the listening angle in relation to the loudspeaker pair very precisely.

Most loudspeaker companies only provide general positioning recommendations, hoping to satisfy a wide variety of different placement preferences of a large number of customers. Necessary for high unit volume products.

As an example of an alternative approach, with ( the clue ), we recommend a very limited use model, with precise placement, and listening angle requirements.

While this is not practical for everyone, due to the restricted use model, it allows us to better optimize the performance of the loudspeaker pair in a way that wouldn’t be possible if we recommended generalized placement.



HOW IMPORTANT IS THIS "COMPENSATION" FOR THE AVERAGE LISTENER?

I’m not sure what you mean by the ‘average listener.’

In terms of amplitude errors related to crosstalk cancellation, they tend to fall in the frequency range that the ear is most sensitive, and are also above just noticeable detection thresholds for amplitude errors, so most any listener with healthy hearing, will detect the difference.

But, your question is valid, in that even though most can discern the difference, will it be important to them?

I would expect discerning listeners to find each change of a similar magnitude to this one to be important, but I guess you would have to decide that one for yourself.



DO WE ALL HAVE DIFFERENT SENSITIVITIES FOR ERRORS IN THIS REGARD?

Interesting question.

I find that each listener comes to a listening session with a personal set of system attributes that they tend to prioritize and focus on in their listening experience and judgment.

BUT, I also find that often in a casual, uncontrolled listening session, listeners will often miss subtle system errors, but if I put that same person in a controlled, blind listening situation, I can almost always teach them to recognize the sonic error, such that after the training, they can then notice the same error in the general listening session.

Again, in the cases of crosstalk cancellation amplitude errors, they appear in the portion of the midrange that the ear is most sensitive to so they will tend to be audible to most listeners.

I hope this all makes sense.

All the best,

- James

[celef]

Celef,

I've put your questions below with answers after...


HOW IS THIS DONE?

There are a number of techniques.

One way is to simply calculate the differential in arrival distance/timing between the left and right channel to one ear. The half-wavelength (and each odd half wavelength) frequency the length differential corresponds to determines the cancellation or dip frequencies from which one can plot the ripple in the response.

Another way is to measure the differential error with a dummy head and in-ear microphones.

Another step is to compare the sound with and without a crosstalk cancelation device, using the crosstalk-cancelled mode as a reference for tonal balance.

Upon establishing the error, it is not necessary to correct all of the resulting measurable ripple. Instead, an gentle spectral correction, on a ½ to 1/3-octave basis, tends to be adequate for this type of error in that the first octave of error has a dip and peak that are a half octave apart. Above the first octave of onset, the comb filtering of the response is closely spaced and has high enough density that it doesn’t require further correction beyond the first octave, or so, of error.

The correction is a delicate one, in that if implemented inappropriately, it can create coloration in the mid-band.

As usual, this is an over simplification, and more is required to deal with the issue effectively, but hopefully this gives you the general idea.

i'm not sure if i understand this, but i sounds close to recent topics about the "bbc-dip" and "fundamental flaw in stereo" we have had at this forum, i couldn't figure out then how this correction should look like, and how to achieve a frequency response that changed dramatically att smal offaxis angles

IF I LOOK AT IT AT DIFFERENT ANGLES I GET DIFFERENT RESULTS FOR EACH ANGLE?

Yes, you are correct. The effect is different at each different listening angle.

That is one of the reasons that one must define the listening angle in relation to the loudspeaker pair very precisely.

Most loudspeaker companies only provide general positioning recommendations, hoping to satisfy a wide variety of different placement preferences of a large number of customers. Necessary for high unit volume products.

As an example of an alternative approach, with ( the clue ), we recommend a very limited use model, with precise placement, and listening angle requirements.

While this is not practical for everyone, due to the restricted use model, it allows us to better optimize the performance of the loudspeaker pair in a way that wouldn’t be possible if we recommended generalized placement.

this is very interesting, how much of the room should be treated as part of the loudspeaker and when starts the room to be just the room? for my own speakers i sometimes use a simple model by roy allison to compensate for boundary gain, even though it calculates this in a simplistic way the compensation often sounds overcompensated

HOW IMPORTANT IS THIS "COMPENSATION" FOR THE AVERAGE LISTENER?

I’m not sure what you mean by the ‘average listener.’

In terms of amplitude errors related to crosstalk cancellation, they tend to fall in the frequency range that the ear is most sensitive, and are also above just noticeable detection thresholds for amplitude errors, so most any listener with healthy hearing, will detect the difference.

But, your question is valid, in that even though most can discern the difference, will it be important to them?

I would expect discerning listeners to find each change of a similar magnitude to this one to be important, but I guess you would have to decide that one for yourself.



DO WE ALL HAVE DIFFERENT SENSITIVITIES FOR ERRORS IN THIS REGARD?

Interesting question.

I find that each listener comes to a listening session with a personal set of system attributes that they tend to prioritize and focus on in their listening experience and judgment.

BUT, I also find that often in a casual, uncontrolled listening session, listeners will often miss subtle system errors, but if I put that same person in a controlled, blind listening situation, I can almost always teach them to recognize the sonic error, such that after the training, they can then notice the same error in the general listening session.

Again, in the cases of crosstalk cancellation amplitude errors, they appear in the portion of the midrange that the ear is most sensitive to so they will tend to be audible to most listeners.

I hope this all makes sense.

All the best,

- James

when looking at measurements of varios loudspeakers at sources like stereophile or soundstage there is hard to detect if they have been compensated for any errors, or at least i have not find any, is this due to the measurement technic they are using or is this compensation rare?

[Jacro]

Celef,

I've put your questions below with answers after...


HOW IS THIS DONE?

There are a number of techniques.

One way is to simply calculate the differential in arrival distance/timing between the left and right channel to one ear. The half-wavelength (and each odd half wavelength) frequency the length differential corresponds to determines the cancellation or dip frequencies from which one can plot the ripple in the response.

Another way is to measure the differential error with a dummy head and in-ear microphones.

Another step is to compare the sound with and without a crosstalk cancelation device, using the crosstalk-cancelled mode as a reference for tonal balance.

Upon establishing the error, it is not necessary to correct all of the resulting measurable ripple. Instead, an gentle spectral correction, on a ½ to 1/3-octave basis, tends to be adequate for this type of error in that the first octave of error has a dip and peak that are a half octave apart. Above the first octave of onset, the comb filtering of the response is closely spaced and has high enough density that it doesn’t require further correction beyond the first octave, or so, of error.

The correction is a delicate one, in that if implemented inappropriately, it can create coloration in the mid-band.

As usual, this is an over simplification, and more is required to deal with the issue effectively, but hopefully this gives you the general idea.

i'm not sure if i understand this, but i sounds close to recent topics about the "bbc-dip" and "fundamental flaw in stereo" we have had at this forum, i couldn't figure out then how this correction should look like, and how to achieve a frequency response that changed dramatically att smal offaxis angles

IF I LOOK AT IT AT DIFFERENT ANGLES I GET DIFFERENT RESULTS FOR EACH ANGLE?

Yes, you are correct. The effect is different at each different listening angle.

That is one of the reasons that one must define the listening angle in relation to the loudspeaker pair very precisely.

Most loudspeaker companies only provide general positioning recommendations, hoping to satisfy a wide variety of different placement preferences of a large number of customers. Necessary for high unit volume products.

As an example of an alternative approach, with ( the clue ), we recommend a very limited use model, with precise placement, and listening angle requirements.

While this is not practical for everyone, due to the restricted use model, it allows us to better optimize the performance of the loudspeaker pair in a way that wouldn’t be possible if we recommended generalized placement.

this is very interesting, how much of the room should be treated as part of the loudspeaker and when starts the room to be just the room? for my own speakers i sometimes use a simple model by roy allison to compensate for boundary gain, even though it calculates this in a simplistic way the compensation often sounds overcompensated

HOW IMPORTANT IS THIS "COMPENSATION" FOR THE AVERAGE LISTENER?

I’m not sure what you mean by the ‘average listener.’

In terms of amplitude errors related to crosstalk cancellation, they tend to fall in the frequency range that the ear is most sensitive, and are also above just noticeable detection thresholds for amplitude errors, so most any listener with healthy hearing, will detect the difference.

But, your question is valid, in that even though most can discern the difference, will it be important to them?

I would expect discerning listeners to find each change of a similar magnitude to this one to be important, but I guess you would have to decide that one for yourself.



DO WE ALL HAVE DIFFERENT SENSITIVITIES FOR ERRORS IN THIS REGARD?

Interesting question.

I find that each listener comes to a listening session with a personal set of system attributes that they tend to prioritize and focus on in their listening experience and judgment.

BUT, I also find that often in a casual, uncontrolled listening session, listeners will often miss subtle system errors, but if I put that same person in a controlled, blind listening situation, I can almost always teach them to recognize the sonic error, such that after the training, they can then notice the same error in the general listening session.

Again, in the cases of crosstalk cancellation amplitude errors, they appear in the portion of the midrange that the ear is most sensitive to so they will tend to be audible to most listeners.

I hope this all makes sense.

All the best,

- James

when looking at measurements of varios loudspeakers at sources like stereophile or soundstage there is hard to detect if they have been compensated for any errors, or at least i have not find any, is this due to the measurement technic they are using or is this compensation rare?

Celef,

Answer to question #1:
I think that the BBC, or “Gundree”, dip is invoked to address audible coloration and inadvertently addresses the crosstalk cancellation errors. If one doesn’t analyze the source of the problem, but instead attempts to correct what is experienced as a tonal error, then one might apply a BBC type dip as a partial fix and hear a sonic preference in the result.

But, if one shapes the response correction to more closely follow the shape of the crosstalk error (not simply a dip), the tonal correction will sound more “life-like”. The BBC dip takes away the “peak” portion of the error, and softens the coloration, but also takes a bit of the “aliveness” out of the midrange.
All of this is fairly subtle, but significant in the totality of the sonic presentation.


Answer to question #2:
If I understand what you are asking, I think of the boundaries that have reflective path lengths within 10 milliseconds, and greater than -10 dB of the direct sound, to be “the loudspeaker”, and the rest to be “the room”.
***
Allison’s work was significant, in that he got people thinking about the importance of considering the effects of the first three boundaries, but it is amazing to me how few loudspeaker companies incorporate Allison’s notions into their loudspeaker design. Roy was also surprised that he had such a small influence on loudspeaker design in the industry.

I’m working on a new paper that I hope to present at AES next year that honors Allison, but shows that his basic thesis that acoustic impedance of the boundary and the change in acoustic impedance due to the distance to each boundary from acoustic matching function dominated the spectral errors. I have found that the issue is not dominated by the acoustical impedance of the boundary, but instead is based on a “specular” reflection differential relationship to the direct sound and this new way of viewing the problem provides a more accurate analysis and correction factor.



Answer to question #3:
I think this type of compensation is rare.

I am not aware of any other loudspeaker manufacturer that claims to use compensation for crosstalk amplitude errors in their loudspeakers. Some may, but I am not aware of it.

As stated above, the most similar approach may be the use of a simple BBC type dip.

Without establishing a very specific listening angle between the loudspeaker and the listener, this approach is difficult to calibrate for correct results and most loudspeaker companies don’t provide strict and specific angular use instructions.

Also, I’m not aware that very many companies design their loudspeakers with any specific design elements that optimize for two-channel usage. Most test and optimize in monophonic mode. Surprisingly, even Floyd Toole and Sean Olive’s “advanced” Harman International loudspeaker test and evaluation center uses only single loudspeaker, monophonic testing and comparative listening sessions. I’ve discussed this with them and, in private, they admit it is a shortcoming that they don’t have a solution for.

For the correction is best to use what I call a “soft correction”, which is a gentle alteration to the frequency response, which may not be easy to interpret from standard measurements. Many loudspeakers have response ripple errors that are as large as the correction, so, if they were applying this type of correction, it could be hard to see the correction hidden within their response errors in the measurements that you have observed.

All the best,

- James

[paa]

Answer to question #1:
I think that the BBC, or “Gundree”, dip is invoked to address audible coloration and inadvertently addresses the crosstalk cancellation errors. If one doesn’t analyze the source of the problem, but instead attempts to correct what is experienced as a tonal error, then one might apply a BBC type dip as a partial fix and hear a sonic preference in the result.

But, if one shapes the response correction to more closely follow the shape of the crosstalk error (not simply a dip), the tonal correction will sound more “life-like”. The BBC dip takes away the “peak” portion of the error, and softens the coloration, but also takes a bit of the “aliveness” out of the midrange.
All of this is fairly subtle, but significant in the totality of the sonic presentation.
All the best,
- James

Interesting, especially the answer to #2 about your upcoming paper.
Anyway, here is previous discussion about the BBC dip:
http://www.faktiskt.se/modules.php?name ... &p=1064224

[single_malt]

Anyway, here is previous discussion about the BBC dip:
http://www.faktiskt.se/modules.php?name ... &p=1064224

Ja, och där framgår den rimligaste förklaringen:

13.I've heard mention of 'the BBC dip' or 'the Gundry dip'. What does that mean?

There is much myth, folklore and misunderstanding about this subject.

The 'BBC dip' is (was) a shallow shelf-down in the acoustic output of some BBC-designed speaker system of the 1960s-1980s in the 1kHz to 4kHz region. The LS3/5a does not have this effect, neither in the 15 ohm nor 11 ohm, both of which are in fact slightly lifted in that region.

According to Harbeth's founder, who worked at the BBC during the time that this psychoacoustic effect was being explored, the primary benefit this little dip gave was in masking of defects in the early plastic cone drive units available in the 1960's. A spin-off benefit was that it appeared to move the sound stage backwards away from the studio manager who was sitting rather closer to the speakers in the cramped control room than he would ideally wish for. (See also Designer's Notebook Chapter 7). The depth of this depression was set by 'over-equalisation' in the crossover by about 3dB or so, which is an extreme amount for general home listening. We have never applied this selective dip but have taken care to carefully contour the response right across the frequency spectrum for a correctly balanced sound. Although as numbers, 1kHz and 4kHz sound almost adjacent in an audio spectrum of 20Hz to 20kHz, the way we perceive energy changes at 1kHz or 4kHz has a very different psychoacoustic effect: lifting the 1kHz region adds presence (this is used to good effect in the LS3/5a) to the sound, but the 4kHz region adds 'bite' - a cutting incisiveness which if over-done is very unpleasant and irritating.

You can explore this effect for yourselves by routing your audio signal through a graphic equaliser and applying a mild cut in the approx. 1kHz to 4kHz region and a gradual return to flat either side of that.


http://www.harbeth.co.uk/faq/index.php#13

Enligt detta var BBC-dipen ursprungligen en åtgärd för att hantera egenskaper hos de högtalarelement som användes. Givetvis har den trots det en psykoakustisk inverkan.

[Objektivisten]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness? Why do I think carbon sounds more real than metal as a conductor? Any clue?

[Jacro]

Answer to question #1:
I think that the BBC, or “Gundry”, dip is invoked to address audible coloration and inadvertently addresses the crosstalk cancellation errors. If one doesn’t analyze the source of the problem, but instead attempts to correct what is experienced as a tonal error, then one might apply a BBC type dip as a partial fix and hear a sonic preference in the result.

But, if one shapes the response correction to more closely follow the shape of the crosstalk error (not simply a dip), the tonal correction will sound more “life-like”. The BBC dip takes away the “peak” portion of the error, and softens the coloration, but also takes a bit of the “aliveness” out of the midrange.
All of this is fairly subtle, but significant in the totality of the sonic presentation.
All the best,
- James

Interesting, especially the answer to #2 about your upcoming paper.
Anyway, here is previous discussion about the BBC dip:
http://www.faktiskt.se/modules.php?name ... &p=1064224

Thank you for the link to your BBC dip discussions. It seems that almost every topic in audio has been discussed in your forum.
I always find enjoyable reading here.

I had some great discussions with Alan Shaw at Harbeth a few years ago. He is a wealth of information regarding the truth about the LS3/5a and BBC loudspeaker research.

It would seem that the LS3/5a is a combination of accuracy and sonic trickery. Odd for a studio monitor to be optimized for sensationalism rather than neutrality. It is a wonderfully entertaining and fun loudspeaker to listen to, but I’m not sure it qualifies as a neutral, professional monitor.

I’ve owned 8 different pairs of LS 3/5a, 15 ohm, 11 ohm, Rogers and Spendors, and still have a couple pair that I enjoy. I had to do a very extensive analysis of the LS 3/5a when I developed the Satterberg Mid-Woofer for them back in the 1980’s. It was very difficult to get a seamless match through the crossover and retain all the sonic attributes of original loudspeaker when used with the Satterbergs.
It was a fun and challenging project, in that we had to develop an unusual approach to subwoofer matching to get it to work properly.

In terms of the BBC/Gundry dip, while it was originally developed to minimize the colorations in the early generations of Bextrene cones, it has been used for many other purposes.

If I remember correctly, I believe the Spendor BC1 and BBC LS3/6 incorporated a Gundry type dip in the upper midrange.

These days this reduction of energy in the upper midrange is most often used to minimize tweeter “flare” or excess off-axis tweeter energy at the crossover frequency.

One of the things that also must be considered, when applying BBC dip type response changes in the midrange, is that they don’t just alter the perception of the tonal character, but they also impact imaging, as the amplitude changes correspond to changes in source angle, as observed in head related transfer functions.

Cheers,

- James

[LypsylateX]

This is one of the best threads in a long, long time. Thank you, all.

(And I just made it a little worse. )

[celef]

james,

thanks for your post, i need some time to think it throu. i have much to ask but one thing comes to mind; this might be against your policy but, what do you think about technologies like www.embracingsound.com

best regards

[DanNorman]

Detta är alltså den raka motsatsen till pip (och guru qm10)?
Ingvar hade problem att hitta en tillräckligt högspridande diskant till pip, medan här väljer man att stoppa den i en waveguide. Intressant.

[DanNorman]

Men givetvis liknande ideer om att integrera högtalaren med rummet.

[sprudel]

Det här är ju jätteintressant!


Citat:

"Although as numbers, 1kHz and 4kHz sound almost adjacent in an audio spectrum of 20Hz to 20kHz, the way we perceive energy changes at 1kHz or 4kHz has a very different psychoacoustic effect: lifting the 1kHz region adds presence (this is used to good effect in the LS3/5a) to the sound, but the 4kHz region adds 'bite' - a cutting incisiveness which if over-done is very unpleasant and irritating."

Så det som jag brukar kalla för överdiven kontrast, eller en analogi till en LCD-skärm med för kraftig skärpeinställning kan vara en höjning i 4kHz området? Vidare skulle en höjning av 1kHz ge en ökad närvarokänsla.

Rörsteg??????

Citat: "A spin-off benefit was that it appeared to move the sound stage backwards away from the studio manager who was sitting rather closer to the speakers in the cramped control room than he would ideally wish for."

Den falska förskjutningen av ljudbilden bakåt! Rymden i ljudbilden?

QLN Signature!

Rörsteg?????

Eller kombinationen av vissa steg med vissa högtalare som ger ett konstlat djup i ljudbilden. Är det så enkelt?

[DanNorman]

Lek med en eq! Det något många skulle lära sig mycket av. Bland annat hur kort ljudminnet är
6db hit och dit.. man vänjer sig snabbt.. sen kan man tro att man hör skillnad på ..... och ........, speciellt utan att blindtesta

[skrutten]

Detta är alltså den raka motsatsen till pip (och guru qm10)?
Ingvar hade problem att hitta en tillräckligt högspridande diskant till pip, medan här väljer man att stoppa den i en waveguide. Intressant.

Kanske att wave guiden ifråga sprider ljudet annorlunda än förväntat?
Bara en tanke... Hursomhelst en intressant tråd detta.

/G

[DanNorman]

Ja, visst. Jag har ingen aning. Bara en tanke.

[Jacro]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness? Why do I think carbon sounds more real than metal as a conductor? Any clue?

Objektivisten;

First, thank you for the friendly welcome. Everyone has been very friendly.

These are big subjects, and often difficult to explore effectively in a short post, but I will start the discussion.

Lately, I tend to avoid answering these types of questions in that I find that many people have already made up their minds and aren’t interested in hearing an answer that contradicts their beliefs.

I have been involved in extensive research in this area, both independently in the early 1980’s and again in the 1990’s when I ran the R&D department at Carver Corporation.

In this post I’ll start with some well-verified, general principles, and if there is further interest I can go into deeper specifics.

One thing that is important to understood is that to find truths about such matters is a complex process and not at all fun if done properly.
Hard work.

It takes a lot more effort and rigor than most audiophiles are willing to endure. For many, audio is a joyful endeavor, and it should be.

It is fun to do casual listening tests and derive and proclaim judgments about many aspects of audio, in terms of what is good, what is bad, all from a casual environment without establishing careful controls and repeating hundreds of tests, over and over again to be sure that one is not fooling themselves.

It is not easy or fun, and I find that there are not many that are willing to put forth the great effort it takes and be open to challenging ones most cherished beliefs.

To me listening to music through an audio system is one of the most fun hobbies one can have. Doing the research to find out which things really sound different, or better, and what really makes a difference, is a completely different kind of, serious, rigorous, activity.

Many seem to confuse the two.

I’m sorry to start by preaching, but I think it is important to establish how careful one has to be to really answer these types of questions with a degree of certainty.

So, I’ll tell you what, after great efforts, I have found to be true.

I’m not saying you should believe me, and I suggest you don’t just believe what anyone says, but test carefully and thoroughly and always find out for yourself. But, be sure to be thorough and rigorous in your testing. Question your most fundamental beliefs as you work through your tests.

Because the subject is so involved, I’ll start with your last question for this first post, and work back to the first question in subsequent posts.

These cannot be complete answers in a single post, but hopefully they will provide a starting point for thought and discussion…and entertainment.

Your Question:

1) WHY DO I THINK CARBON SOUNDS MORE REAL THAN METAL AS A CONDUCTOR?

First, I’m not sure why you think that, as I don’t know how you have tested your hypothesis to arrive at your conclusion.

So maybe the question is not, "Why do you think carbon sounds better than metal?", but maybe your real question is, "Is it true that carbon sounds more real than metal?"

If my answer about whether it is true or not, doesn't satisfy, then we can explore the psychology and process of "Why do YOU think it is true?"

I have found that one must not make judgments about any component out of the context of the system that it is used in. It has almost no meaning to discuss component types outside of the circuit that they operate in, because the quality of their performance is dominated by the relational impedance interface within the system.

You might say, I have found this system (amp, pre-amp, loudspeaker, etc.) that when I use resister “A”, it sounds better to me that when I use resister “B”. You may even say, that I have found this pattern in a number of systems.

I would tend to ask first, what is it about these systems that they react in an unfavorable manner when used with resister “B”? Is the circuit interface compatible with resistors that are of type “B”? Is resistor type “B” being used in the manner that it is best suited?

While it might seem more obvious to question the component “resistor type B”, it turns out, that unless one has a component that is fundamentally defective, or is functioning outside a standard deviation of some important parameter, it is more likely that what you are experiencing as a sonic difference, is the system mismatch, not the inferiority of the component in isolation.

Sometimes one will find that in a certain component category, such as a capacitor, the one that is the “audiophile favorite” is not necessarily the best to use in a particular circuit.

At a recent show, people kept asking me what kind of capacitors I used in the crossover of ( the clue ) loudspeaker. I would have to say, that I cannot give a simple answer to the question.

I have a number of networks designed that all sound the same. Each network has different components from different vendors.

The reason I had to design different networks is that I have certain components that I cannot second source, so if they become unavailable, I have to substitute another component from another vendor.

But, in some cases, another component will not work perfectly, just substituted in, even if it has the same specified values. The transfer function will have changed a little. So, when I have to change one component, such as a capacitor from a one brand name to one of another, I may have to change the components that interface the input and/or the output of that capacitor to maintain the same dynamic transfer function, and sound quality. One cap wasn’t better than the other, but the system that it was operating within, was better matched to achieve my ideal input to output function.

So, to be able to use alternative components in certain places in the crossover network, I designed different networks to best work with different component set combinations while maintaining the same dynamic transfer function.

When I first substituted the replacement part, and found the system to perform worse, I could have blamed the quality of the part and decided not to use it, and call it “a bad part”. But, it wasn’t “a bad part”, it was a part that the system was not matched to. Once properly re-matched with new associated components, one could not hear any difference.

It can easily happen that substituting the most expensive component will actually disturb the transfer function of the total system. Sometimes it will sound different, when put in the circuit, and listeners will assume the difference means “better” because they think it is a better, more expensive component, that has a good reputation, but it doesn’t always work that way.

Good audio is predominately a systems design approach, not an individual component selection approach. The components have to be appropriate within a standard deviation, but beyond that, it is all based on the impedance interfacing of the system.

Some designers operate from a systems approach, and some try to use all the most expensive components and hope for the best result.

Whether one has an unlimited budget or is designing for lowest cost, the systems approach will provide a better result and will tend to be less wasteful. But, it takes longer and requires more careful assessment.

Cheers,

- James

[Kraniet]

Im a little bit more interested in the waveguide. I see you use a very small WG, so small that some people (like geedle) dont even call it a waveguide.
However I think reasonably sized WGs are more interesting and also would have more relevance in home-audio (as its more "design-driven").

Ive been looking at JBL* and what they call "elliptical oblate spheroidal waveguide" and by the looks of it it seem to be only 5x8cm or something like that. Thats very small when it comes to waveguides but they (JBL) seem happy with the directivity control, or matching, they get.
Looking at the site you linked to, small waveguides seem rare.

Do you fell that the WG have to be big to be effective? If one assumes a 2kHz crossover, is a size like JBL WG enough or is it just for show?

The DXT-lens is a diffraction lens and maybe not a true WG, but it is very effective att directivity control down to 2Khz. But being a diffraction-device there should be alot of HOM being created. Seeing the "rave" that this lens have got (like in the speakers from Acoustic Energy) it would seem that HOMs arent that bad?

In what way are you using the WG in your speaker and isnt it very short and of a "simple" profile to be effective as a WG? (no disrespect intended of course)


Heres the link to JBLs Linear Spatial Reference Studio Monitor System

*http://www.jblpro.com/catalog/general/Product.aspx?PId=26&MId=5

[Laila]

. . . . . .

Good audio is predominately a systems design approach, not an individual component selection approach. The components have to be appropriate within a standard deviation, but beyond that, it is all based on the impedance interfacing of the system.

Some designers operate from a systems approach, and some try to use all the most expensive components and hope for the best result.

Whether one has an unlimited budget or is designing for lowest cost, the systems approach will provide a better result and will tend to be less wasteful. But, it takes longer and requires more careful assessment.

Cheers,

- James

James, welcome to the forum !

Like poetry in my ears(eyes) . . . . wounderful words.

[Jacro]

james,

thanks for your post, i need some time to think it throu. i have much to ask but one thing comes to mind; this might be against your policy but, what do you think about technologies like www.embracingsound.com

best regards

Celef,

I look forward to your next questions.

In terms of the Embracing Sound System, I think it is a very promising technology.

The basic principle of an crosstalk cancellation systems have has been attempted and explored since the mid 1960’s.

In 1963, Schroeder and Atal out-lined a method for generating a phantom source. These ideas were elaborated by Damaske in 1971 and were followed in the mid-1970’s by Iwahara in Japan, with the JVC’s Bi-Phonic processor. As the pure theory suggests, all were made for use with Binaural recordings. In 1979 Robert Carver created a modified version of the JVC unit and offered a processor recalibrated for stereo recordings called Sonic Holography.

All were attempts to fix some of the fundamental spatial and tonal flaws of 2-channel stereo. Theoretically, under ideal conditions, it can be a superior approach, but is much more critical to optimize than conventional stereo. While potentially better than stereo, if it isn’t perfectly calibrated it can sound much worse than stereo. Ultimately, it is best suited for reproducing a binaural based recording.

These types of systems were always problematic with widely spaced loudspeakers, because the cancellation signals were very difficult to match, as they had to include the frequency response effects of the sound rapping around the face, to the opposite ear.

Around the same time (about 1980-1) we built a crosstalk isolation wall/barrier with a loudspeaker on each side of the barrier, with the wall projecting from the loudspeakers to the face of the listener. The spatial effects were quite impressive, and formed a reference sound field for further development work but the structure was not practical for consumer use. (Don Keele wrote an AES paper on this technique in 1986) With this structure as a reference, we worked on the processor approach and found that if we placed the loudspeakers close together that the correction for face-related-transfer-functions were much easier to apply and retain neutral tonal balance, and the spaciousness was even more impressive.

Unfortunately, due to the new movement towards multi-channel Surround Sound becoming popular, our two-channel project was halted and 5-speaker surround sound was where development efforts were redirected.

In 1995 new work began by Kirkeby, Takeuchi and Hamada, (University of Southhampton and Tokyo Denki University) as filed in British Patent App. # 9,603,236.2 “Stereo Dipole”. This closely spaced cross talk canceller was found to have lower coloration and greater stability of image with head movements. The same group wrote a series of very interesting papers on the subject of Stereo Dipoles, provide thorough analysis and synthesis information that was much more advanced than the work we did in the 1980’s.

It appears that the Embracing Sound Systems are based on the work of Kirkeby, Takeuchi, and Hamada (I don’t think their patent was ever granted) and if properly implemented, should be very good.

It has always been a system with great potential, particularly with binaural recordings. While it can provide an impressive demo with standard stereo recordings, it doesn’t reproduce an ideal transfer from stereo recordings, at least not as an accurate representation of the source, but it is a lot of fun to experiment with.

Now if we could get all the recording companies to offer a binaural alternative to every stereo recording, life would be great, but I don’t think that is going to happen anytime soon.

So, in the meantime, those of us that design reproduction systems, must work blind to the recording process, never knowing what kind of microphone technique is going to be used. Ultimately, an impossible task, but we persevere, searching for the best ways to create a believable facsimile of the original sound from an imperfect model.

All the best,

- James

[paa]

James,
Maybe you have a comment to Professor Edgar Choueiris work also?

http://www.princeton.edu/3D3A/

[paa]

I have been working on new waveguide topologies that are based on an elliptical variation of the Oblate Spheroid.
- James

Seems like Procella has gone towards that direction too (and it's smaller than it looks in the picture):

[Lust]

Here we have some larger wave guides ...

http://www.gedlee.com/

Experience/comments someone?

[Jacro]

Im a little bit more interested in the waveguide. I see you use a very small WG, so small that some people (like geedle) dont even call it a waveguide.
However I think reasonably sized WGs are more interesting and also would have more relevance in home-audio (as its more "design-driven").

Ive been looking at JBL* and what they call "elliptical oblate spheroidal waveguide" and by the looks of it it seem to be only 5x8cm or something like that. Thats very small when it comes to waveguides but they (JBL) seem happy with the directivity control, or matching, they get.
Looking at the site you linked to, small waveguides seem rare.

Do you fell that the WG have to be big to be effective? If one assumes a 2kHz crossover, is a size like JBL WG enough or is it just for show?

The DXT-lens is a diffraction lens and maybe not a true WG, but it is very effective att directivity control down to 2Khz. But being a diffraction-device there should be alot of HOM being created. Seeing the "rave" that this lens have got (like in the speakers from Acoustic Energy) it would seem that HOMs arent that bad?

In what way are you using the WG in your speaker and isnt it very short and of a "simple" profile to be effective as a WG? (no disrespect intended of course)


Heres the link to JBLs Linear Spatial Reference Studio Monitor System

*http://www.jblpro.com/catalog/general/Product.aspx?PId=26&MId=5

Thou shalt not judge a man by the size of his waveguide
- Anonymous Swedish philosopher

A waveguide doesn’t always have to be big to be effective. Generally, the purpose of a waveguide is to increase the directivity of the tweeter to more closely match the directivity of the woofer for frequencies that are at, and near, the crossover. To achieve that purpose, the waveguide should be comparable in dimension to the active radiating portion of the woofer.

Also, generally, one wishes to maintain the smallest center-to-center distance between the woofer and the tweeter, to avoid interference nulls in the vertical response.

So, if one had a 160mm woofer, with a tweeter attached to a 300mm waveguide, the overly large size of the waveguide would be problematic in at least two different ways:

- The tweeter/waveguide would have significantly more directivity at the crossover frequency than the woofer, causing a mismatch and very poor off-axis frequency response

- The center-to-center spacing between the woofer and the tweeter would be quite large (at least 230mm) causing a very narrow vertical listening window between the vertical nulls in the polar response

An additional point is that is significant, is that it is difficult to determine the radiating size of a given woofer just by measuring the size of the diaphragm. Some woofers and midranges will have a smaller source size with increase in frequency, due to the outer portion of the cone breaking up and decoupling from the center portion of the cone. So the waveguide may need to be a bit smaller than the cone diameter to have matched directivities at a given frequency.

Big woofers = Big waveguides.

Small woofers or midranges = Small waveguides.

With the JBL link that you provided, the crossover to the tweeter/waveguide is stated as 2.2 kHz. The waveguide appears a little small but again, it may be size-matched to the characteristics of the midrange. JBL considers off axis frequency response, and directivity matching, to be one significant parameters, so one can assume they chose their waveguide size very carefully.

The DXT waveguide does have diffraction steps but they are fairly shallow and hopefully don’t develop significant amounts of high order modes, but I am not certain what amount of HOMs are created in the DXT waveguide.
I would expect that the very good directivity control of the DXT waveguide when combined with a high quality transducer will produce good results.
It would be interesting to add a foam insert to absorb the high order modes and find out if eliminating the HOMs significantly improved the sound quality.

After extensive testing, the form and size of the waveguide in ( the clue ) was chosen to achieve the desired effect in matching the interaction with the cone driver over the transition frequencies. If we can make the system perform better by making the waveguide larger, or of a more complex form, we would do so, but so far, we have found the current waveguide to be the best match.

(No disrespect taken )

- James

[Kraniet]

Thats reassuring to hear. Listening to Geddes one get the feeling that everything goes as long as its by his defenition..

You mentioned Revel and JBL earlier and their work on evaluating meaningful speaker attributes. I think some of it we could all agree to, like a reasonably flat on-axis and a even and slowly falling off-axis for higher frequencies. But their testing method seem limited to "voicing" and percieved sonic balance. The issues with the stereo flaw doesnt seem to enter the picture.

Why is that do you think? Is it a little to "subjective" of a thing for them to be able to "double blind" it? As you say they seem aware of their limitations.

But isnt it also a good way to go, establishing the "true objective" attributes of speakers like frequency/power response, low distortion etc? I mean looking at the industry at a whole there seem to be a lack of an "even ground" for how a speaker should reasonably work for it to have a chance of being a true reproducer och the source material. Lacking those attributes it seems useless to optimize for stereo imaging. Or?

[Jacro]

James,
Maybe you have a comment to Professor Edgar Choueiris work also?

http://www.princeton.edu/3D3A/

paa,

The work of Dr. Choueiri is very impressive, and encouraging, in its thoroughness.

He has taken the cross-talk cancellation process beyond Local simplicity and has included the Global issues, such as listener body related transfer functions and the room issues. His use of the power of digital signal processing to address the secondary effects of XTC is what makes his system more encouraging than the work that has been attempted by others.

There is still the issue that post processing of this type is only “correct” for binaural recordings, but as he suggests, it will also work reasonably well for any stereo recording that has attempted to preserve the original inter-aural time and level differences.

Choueiri is one of the few that has suggested, and required, directivity control in the loudspeakers used with his system. He addresses how the room effects distort the correction signals and he has developed a very sophisticated filter system to address the tonal coloration found in all the prior art systems of this type.

His custom filter calibration to listener location and attributes is also a step forward.

I have been following his work since he first made it public, and I have not yet measured or heard one of his systems. But the fact that he is not just talking about the difficulty of secondary problems, and instead is actually addressing them is very encouraging.

Theoretically, he appears to be on the right track in many ways.

Cheers,

- James

[Kraniet]

seems limited to one person listening though. he speaks of it being viable for TV but I cant see how it would work in a normal living room with 4-6 people getting the full experience.

[Jacro]

seems limited to one person listening though. he speaks of it being viable for TV but I cant see how it would work in a normal living room with 4-6 people getting the full experience.

Yes, your observation and concerns are correct.

If one moves their head, even though the Stereo Dipole is more robust than other XTC systems, it still tends to be limited to an ideal listening region of less than half a meter square...Not enough for more than one listener.

Currently, the best 2-channel approach for multiple listeners is to use a conventional stereo system, incorporating time/intensity trading, with the most intense axis of the loudspeakers crossing in front of the listener, as originally proposed by Ben Bauer in the 1960s and Mark Davis at DBX in 1983 (US 4,503,553).

Modern versions of this approach use waveguides for directivity control and cross fire the loudspeakers in front of the listeners.

As a general rule though, the potential of the accuracy of an audio signal at the listening seat is inversely proportional to the size of the listening region around the head.

So, the best sound is always in the center, but the best cross axis solutions can make the sound quality remain quite satisfying a few people sitting together.

Best regards,

- James

[Jacro]

I have been working on new waveguide topologies that are based on an elliptical variation of the Oblate Spheroid.
- James

Seems like Procella has gone towards that direction too (and it's smaller than it looks in the picture):

Yes, a number of groups are starting to work with this form of waveguide, as it is found to be quite effective in regards to consistent directivity control, low linear/diffractive distortions, and allows reduced center-to-center spacing compared to axi-symmetric waveguides.

Some, such as Geddes, point out that it doesn't match the directivity of the midrange/woofer in the vertical axis as well as an axi-symmetric waveguide and they believe that the center-to-center spacing advantage is not as important as directivity matching.

The counter argument is that while the local directivity of an axi-symmetric waveguide and woofer, of approximately the same size, DO match better at the crossover frequency, the Global, combined vertical directivity of the two symmetrical units causes a significant directivity change at the crossover frequency.

And the pro- axi-symmetric group claims they can control that effect in the crossover... and... and...

Proper system design can result in great performance from either approach, with one offering some advantage over the other depending on all the system variables.

- James

[Jacro]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness?

Hi Objektivisten,

I still owe you an answer on your Tube amp and Power requirement questions.

I’m still not sure how to address such vast topics adequately in a single post.

So, to start, I’ll provide you with some information that you may find interesting from earlier research that we did, and I will spread the discussion over a few posts.


Part One (A partial review of Large Signal/Overload Issues):

1) WE FOUND THAT PARTIAL CLIPPING OF AN AMPLIFIER IS NORMAL USAGE FOR MANY LISTENERS.

- We monitored the systems of 92 different audiophiles over a 2-year period by monitoring their amplifiers in their homes and collecting data on how often their amplifiers were clipped.

When using high quality program material, we found that over 70-percent of the test subjects used their systems in a manner that regularly caused the onset of overload, or mild clipping, even though most of them would not be considered to listen at inappropriately loud levels.

(We eliminated the data of the subjects that obviously drove their systems beyond their capability and into hard clipping.)

(The average rated wattage of their amplifiers was 45 watts per channel, with the lowest being 9 watts and the highest being 350 watts per channel. Average rated loudspeaker sensitivity was 88dB/1 watt.)

2) WE FOUND THAT SOME AMPLIFIERS HAVE EUPHONIC CLIPPING CHARACTERISTICS THAT CAN ACTUALLY ENHANCE SOUND QUALITY DURING MILD CLIPPING LEVELS.

- We found that there is a wide variety of overload parameters and associated audible artifacts that are derived from those overload parameters.

• Harmonic distortion: Depending on whether the amplifier generated predominately low order*, even harmonics*, or high order**, odd harmonics** determined whether the clipping sounded “enhanced*” or “irritating**”.

Many tube amplifiers generate low order, even harmonics during overload, due to their asymmetrical clipping characteristics.

Most solid-state amplifiers produce primarily odd-order harmonics due to their symmetrical clipping characteristics.

• Other characteristics that can provide either ‘euphonic’ or ‘irritating’ sound quality during overload;

Overload In-phase or out-of-phase cross talk:
(Out-of-phase cross-talk can cause enhanced spatial effects. Some tube amplifiers exhibit this characteristic during mild overload)

Overload recovery time:
(Can prolong either positive or negative overload effects, unless recovery is extended due to “sticking” wherein an amp will stay “stuck” to one power supply rail is itself a malady of the design)

Overload output impedance increase:
(Can cause under-damping of loudspeaker during overload. High feedback amplifiers can have a greater delta between linear and overloaded output impedance, which may cause a more disturbing sonic result.)


3) SOME HIGHER POWERED AMPLIFIERS ARE CRITICIZED FOR POORER SOUND QUALITY WHEN COMPARED TO LOW POWERED AMPLIFIERS:

- It turned out, that in nearly every case, the higher-powered amplifier was not inherently worse sounding, but because it had more unclipped power, it was found that it could more easily over-drive the loudspeakers into greater non-linearity, causing the loudspeaker (not the amplifier) to have worse sound.

But since the amplifier was the only element changed in the system when comparing the sound, the higher-powered amplifier was most often unfairly blamed for the decreased sound quality.

4) In later studies we have found that many class-D amplifiers have an even more severe problem in that they start to have significantly increased distortion effects as they approach clipping, but have not yet overloaded. Because of this, they may not be able to be used in a manner that fully utilizes their full power capabilities without sonic changes.


So, in conclusion:

A - Hard clipping is very irritating in any amplifier, but mild onset clipping can provide a wide variety of sonic effects, some of which can actually be experienced as “more enjoyable” than unclipped. Many tube amplifiers exhibit similar overload characteristics that may be experienced as “euphonic”.

This effect of mild overload, being sometimes experienced as favorable, is partially due to the fact that program peaks tend to be very brief and the ear’s ability to accurately analyze the quality of the peak energy is greatly reduced and more easily fooled as compared to the ear’s ability to evaluate long-term average levels.

B – Larger amplifiers can easily be blamed for poorer sound quality if they are used with loudspeakers that change sound quality when driven to the greater peak levels of the larger amplifier, even when the average levels are held to the same as comparable low powered amplifier.

Some loudspeakers have such low levels of linear capability that merely changing from a 10-watt to a 35-watt amplifier may cause this confusing effect of the larger amplifier being the cause of the sonic deterioration.

Again, this is a very limited discussion of the topic, but it may possibly explain experiences that some of you have had with differences between power amplifiers.

If there is further interest, I’ll try to follow up soon with a bit more information on the topic, including a few small signal issues with power amplifiers.

I originally interrupted your discussion here just to answer a few loudspeaker questions, and I don’t want to hi-jack your forum, so let me know if I am inappropriately dominating your discussions.

I would be curious to see what everyone else’s thoughts are on some of these subjects, and/or what everyone considers to be the largest remaining questions and areas of concern with their audio systems.

Best regards,

- James

[Jacro]

. . . . . .

Good audio is predominately a systems design approach, not an individual component selection approach. The components have to be appropriate within a standard deviation, but beyond that, it is all based on the impedance interfacing of the system.

Some designers operate from a systems approach, and some try to use all the most expensive components and hope for the best result.

Whether one has an unlimited budget or is designing for lowest cost, the systems approach will provide a better result and will tend to be less wasteful. But, it takes longer and requires more careful assessment.

Cheers,

- James

James, welcome to the forum !

Like poetry in my ears (eyes) . . . . wonderful words.

“A waveguide by any other name would sound as sweet”


Hmmm… Maybe it is best that I not pursue a career as an audio poet, and just stay focused on the technical discussions -- But thank you for your kind words.

- James

[Stereotypen]

Hi, and Welcome Jacro!
Very interesting ta read about your study of amplifiers/loudspeakers!
I am interested to hear what you think about Siegfried Linkwitz approach to loudspeaker design, dynamic dipoles.
I myself have found them to solve a great deal of the problems in the speaker/room area in my listening room. The uniform powerresponse may be part of the answer here i guess.
What are your experiences or oppinions regarding this?

[Laila]

If there is further interest, I’ll try to follow up soon with a bit more information on the topic, including a few small signal issues with power amplifiers.

Please do that. . .

I originally interrupted your discussion here just to answer a few loudspeaker questions, and I don’t want to hi-jack your forum, so let me know if I am inappropriately dominating your discussions.

Not a chance(I think ) !

[Nattlorden]

Jacro - almost no thread here ends on the same note as they begin, so that's completely natural. And feel free to comment/contribute in other threads than this one - I think I speak for everyone here when I say we're delighted to have yet another professional designer on the forum.

It's interesting when you all agree and equally so when you don't....

[Jacro]

Hi, and Welcome Jacro!
Very interesting ta read about your study of amplifiers/loudspeakers!
I am interested to hear what you think about Siegfried Linkwitz approach to loudspeaker design, dynamic dipoles.
I myself have found them to solve a great deal of the problems in the speaker/room area in my listening room. The uniform powerresponse may be part of the answer here i guess.
What are your experiences or oppinions regarding this?

Hi Stereotypen,

I have great respect for the work that Linkwitz does and the generosity he has come to exhibit by sharing so fully on his website. (He was much more guarded with information in his younger days.)

You are correct in assuming the importance of uniform, or at least carefully tailored, power response.

My general recommendation is that once one has established an effective direct sound, with minimum interference from early correlated reflections, then the ideal is to have the best full range balance possible on all secondary room sounds, preferably with the appropriate delay between the direct sound and correlated reflections.

To achieve this, I believe the loudspeaker should either be operated as a boundary coupled half-space device, or as an omni or full range dipole well out into the room.

(These out-in-the room, symmetrical dipole, systems work best in larger rooms, so as to be able to be well removed from the front wall, and sidewalls. Small rooms do not accommodate them very well, particularly the more omni-directional devices. Dipoles can be oriented to aim the dipole nulls at the sidewall and front wall first reflection points to minimize early reflections to help work better in a smaller room, but that is still a compromise.)

Unfortunately, most loudspeakers operate in the awkward middle ground between these two approaches. Most (cone in a box) loudspeakers are operated a short distance away from the front wall, but they have omni-directional response at all lower frequencies, and more directional sound in the upper frequencies, with “uneven” power response creating an unbalanced “room sound”. The “rear 180-degree wave angles” tend to exhibit very uneven, frequency response and drive the room to an unnatural secondary sound field.

While I feel the half-space, boundary-coupled approach is the more universally consistent way to achieve the best effect, the Linkwitz approach can also provide a very good result, particularly if the room size is compatible. The equal front/back response at least drives the room in a manner that is consistent with the direct sound, but there are some questionable side effects, such as artificial reflections off the front wall behind the loudspeakers, often creating interference with the direct sound, and also creating “false depth”.

False depth can be an enjoyable effect, but it does not represent an accurate mapping of the original event into your room.

Only recently has Linkwitz finally arrived at making sure that the dipole response is equal on the front AND backside of the loudspeaker. Previously, he only operated dipole up to approximately 2kHz, but now he has adopted full symmetry.

If one is putting a loudspeaker out into the room, it must be either very symmetrical, preferably with constant directivity vs. frequency, and even better, constant “high” directivity with frequency, or it must be power response tailored in a very careful manner.

A few loudspeakers that don’t follow the guideline (of symmetry or half-space), can be made to perform well, but must be tailored very carefully to be successful (Quad ESL 57).

Otherwise, to work well, they should be used in the near field, eliminating room sound as much as possible (Harbeth HL-P3, LS3/5a).

Again, all of this is an over simplification, but as a guideline, but hopefully, it at least points to some of the issues that must be dealt with.

- James

[paa]

...While I feel the half-space, boundary-coupled approach is the more universally consistent way to achieve the best effect,-
...
James

You really sound like the Carlsson-followers here. Have you studied Stig Carlssons loudspeakers to any extent?

[Jacro]

...While I feel the half-space, boundary-coupled approach is the more universally consistent way to achieve the best effect,-
...
James

You really sound like the Carlsson-followers here. Have you studied Stig Carlssons loudspeakers to any extent?

I suppose I do sound somewhat like the Carlsson followers.

And, yes, I looked at his speakers, paper, and patents quite a few years ago. I believe that at least part of his concept that he developed in later years was valid and useful.

Carlsson, Snell, and Allison had all gotten part of the puzzle solved.

I'm not saying one can't get enjoyable sound with a non-boundary coupled loudspeaker, but it is much more difficult to create an accurate representation of the original event if one stimulates early, correlated front wall reflections, and do so with a spectrally distorted wavefront from the back and sides of the loudspeaker.

Boundary coupling makes things easier by creating a simpler wave-launch architecture and taking care of a number of issues that are normally rather difficult to deal with.

- James

[Kraniet]

Peter Steindl on this forum seem to draw on all those speakermakers (and some others im sure) in his "Egg"-speaker http://www.bremen.se/main/HT12.0/HT12.0.html
He speaks of his background and a little about the theories behind his speakers in this thread http://www.faktiskt.se/modules.php?name ... ic&t=16989
One claim being true 3D-sound with just 3 speakers. Its all very interesting, although its a bold claim.

Ive been thinking in terms of wall-close speakers but its not all that easy. The shallow, flat box arent always that easy to position along the wall with the wall-sockets and what have you. So its either relatively small boxes hung on the wall or more conventional florstanders made for a close placement against the wall.

Older Carlsson-speakers suffer alot by the fact that they are spreading the sound everywhere. Makes it all very diffuse and an artificial "soundstage". I call it painting with a wide (to wide) brush. The later madels work better but need a substantial dampening on the wall to not get that diffuse soundstage.
I get the same feeling with dipoles. But i guess some well placed damping and diffusion can work miracels with a dipole.

[Stereotypen]

Thank you Jacro for answering my question and sharing your thoughts.
I hope you will find this a pleasent forum and decide to drop in here every now and then.

[paa]

James, when you try to get a rectangular box like The Clue as coupled as possible to the wall, how much absorbtion would you prefer behind it? What size and thickness would you consider minimum and optimum?

[Jacro]

Peter Steindl on this forum seem to draw on all those speakermakers (and some others im sure) in his "Egg"-speaker http://www.bremen.se/main/HT12.0/HT12.0.html
He speaks of his background and a little about the theories behind his speakers in this thread http://www.faktiskt.se/modules.php?name ... ic&t=16989
One claim being true 3D-sound with just 3 speakers. Its all very interesting, although its a bold claim.

I had previously read about the Bremen loudspeakers on Faktiskt, but I do not remember seeing a 3-speaker version discussed.

The 2-speaker version of the use-model concept seems to be very much the same as the Harold Beveridge line source electrostatic from the 1970's, wherein Mr. Beveridge boundary coupled his loudspeakers to the side-walls, about 1/3 of the distance along the side-wall, facing each other. They were linesource, and electrostic, but the general sound field presentation seems the same.

http://www.bevaudio.com/
http://bevaudio.com/technical_details.html

The perceived sound field from this type of presentation is quite impressive, spacious and deep, and fun, but some other aspects are questionable.

Also, it is difficult to find the correct room for them, that has the optimum length to width ratio.

They were the most expensive loudspeakers when they were first available. I've always thought it would be fun to have a low cost conventional-driver version that would create the same sensationalized sound field. I built prototypes years ago of planar magnetic line sources arranged in this manner and it worked quite well. Very entertaining.

It would seem that the Bremen speaker arrangement may provide this effect very well, and in a cost effective manner, but I have not had the opportunity to hear them.

Ive been thinking in terms of wall-close speakers but its not all that easy. The shallow, flat box arent always that easy to position along the wall with the wall-sockets and what have you. So its either relatively small boxes hung on the wall or more conventional florstanders made for a close placement against the wall.

Yes, if a loudspeaker is specified for "wall-close" or corner placement, or any specific placement, then, unfortunately, there are always a significant number of people that don't have that kind of space available in their homes. That is the problem with specified placement, but the advantages of designing to a specific placement are so compelling, that I believe that I must choose that path, even if a majority of potential customers choose to avoid the product because they don't have a proper place for it.

There are plenty of other options for those particular customers and hopefully they can find a loudspeaker that makes them happy, in placement, appearance, and sound quality.

Older Carlsson-speakers suffer alot by the fact that they are spreading the sound everywhere. Makes it all very diffuse and an artificial "soundstage". I call it painting with a wide (to wide) brush. The later madels work better but need a substantial dampening on the wall to not get that diffuse soundstage.
I get the same feeling with dipoles. But i guess some well placed damping and diffusion can work miracels with a dipole.

Ultimately, to get the best performance from each system type, one needs to be able to provide best placement and the associated room treatment that supports that particular system type.

The only system architecture that I have found to substantially transcend those requirements is to build a highly directional loudspeaker that inherently "ignores" the room for all early reflections. This type of hyper-directivity is very difficult to achieve in a practical manner, over a wide bandwidth, but the net result can be very impressive and transportable to a wide variety of acoustic environments.

I developed these types of systems from about 1994 to 2008, with some of the systems using ultrasonic to audio conversion (a parametric loudspeaker), and while they were not particularly practical, they could be very impressive in their ability to eliminate what I call the "Second Venue Problem", allowing one to hear the original, or "First Venue", where the original performance was recorded in a very convincing manner.

All the best,

- James

[Jacro]

James, when you try to get a rectangular box like The Clue as coupled as possible to the wall, how much absorption would you prefer behind it? What size and thickness would you consider minimum and optimum?

As a minimum I recommend 75mm thick acoustic foam that is arranged on the wall, inward from the right loudspeaker (going towards the left loudspeaker), starting at the inside corner of the cabinet that is closest to the wall, and also starting at the bottom edge of the cabinet.

The panel should be about 60 x 60cm, extending about 25cm above the top of the cabinet.

Ideally, additional amounts and placement would be adapted to the needs of a particular environment.


For a more complete/optimal arrangement, as a general rule, I would suggest a 75mm thick, 150cm x 150cm square piece centered horizontally behind the cabinet, and starting at the floor.

Also, place 100mm thick acoustic foam outward from the speaker, (to the right of the right loudspeaker) over to the corner, and from the corner extend 1/3 the length of the sidewall, OR, a piece 1-meter wide centered on the point where the first specular reflection bounces off the sidewall to the listener.

Again, each room has it’s own issues to deal with, but these are the general guidelines for minimum and optimal absorption.

Let me know if I need to provide further explanation.

Cheers,

- James

[Jacro]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness? Why do I think carbon sounds more real than metal as a conductor? Any clue?

Objektivisten,

I was just curious... I have not seen any comment or feedback from you on the answers I provided so far to your questions.

It is a rather complex topic, so I thought I should check to see if my starting answers were useful for you, such that you would like me to continue, or not?

Best regards,

- James

[Objektivisten]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness? Why do I think carbon sounds more real than metal as a conductor? Any clue?

Objektivisten,

I was just curious... I have not seen any comment or feedback from you on the answers I provided so far to your questions.

It is a rather complex topic, so I thought I should check to see if my starting answers were useful for you, such that you would like me to continue, or not?

Best regards,

- James

Thanks for a very interesting an thoughtfull answer James. I have wait in silence and hope for the other topics, and I have read all your inputs in this thread and understand You have a had a lot to do. Please welcome.

[Kraniet]

[quote="Jacro"][/quote]

The Bremen "Egg" is supposedly able to work with just thw two speakers, but as I understand a third, "center", speaker is being developed.

It would be interesting to hear your opinion on the multi-channel audio. There is a few people saying that the best way to get "true stereo" is by using three speakers, thus bypassing the "stereoflaw". There are a few processors doing multichannel from stereo, like dolby PL och Harmans Logic 7. Do you think its a good idea trying to recode stereo into multichannel? True multichannel recording might be better but were seeing some pretty stupid example of instruments panned hard to the rear sides etc.
Mr Steindl have is own "black box". Are the existing systems not good, and why (in your opinion).

regarding the placement of speakers its true like you say. But the best thing for a loudspeaker developer would be to make speakers that fit most persons/rooms. Your Clue for instance seems a better choice than say a Snell type A.
In my case its was only when I tried to position my 45cm wide speakers that I realized they werent as placement-friendly as I thought. With the type A´s it seems youd have to modify the room after the speakers to be able to position them correctly.
Making the speaker more like your Clue seems a better choice i that case.
Did you consider an even more "wall-integrated" desing like the Carlsson OA51 or Larsen speakers? http://www.larsenhifi.com/larsen8.htm

[sportbilsentusiasten]

Jacro - almost no thread here ends on the same note as they begin, so that's completely natural. And feel free to comment/contribute in other threads than this one - I think I speak for everyone here when I say we're delighted to have yet another professional designer on the forum.

It's interesting when you all agree and equally so when you don't....

Well stated, agree 100%!

Welcome Jacro. Even if this is my first response in the thread, you should know that I enjoyed immensely your contribution to the forum.
I know many other people that read a lot more than post...

On this forum we just LOVE discussing technical details, especially based on own experience and studies.
Theory and calculations are interesting subjects in itself but nothing beats personal experience imho.

I'm going to respond to a few of your posts later on.
To be continued as they say...

[sportbilsentusiasten]

To achieve this, I believe the loudspeaker should either be operated as a boundary coupled half-space device, or as an omni or full range dipole well out into the room.

I agree on pretty much everything in this post and have a Q regarding this quote - please list a few of your favorite speakers during the years

Yes I read about your reluctance commenting on existing products so list discontinued speakers then.
(even if I consider giving positive feedback about a competing product something that gives cred )

[sportbilsentusiasten]

The 2-speaker version of the use-model concept seems to be very much the same as the Harold Beveridge line source electrostatic from the 1970's, wherein Mr. Beveridge boundary coupled his loudspeakers to the side-walls, about 1/3 of the distance along the side-wall, facing each other. They were linesource, and electrostic, but the general sound field presentation seems the same.

http://www.bevaudio.com/
http://bevaudio.com/technical_details.html

The perceived sound field from this type of presentation is quite impressive, spacious and deep, and fun, but some other aspects are questionable.Also, it is difficult to find the correct room for them, that has the optimum length to width ratio.

Most (if not all) speakers are compromises so please elaborate about the not so great aspects please.

What is according to you the best length to width ratio?
Is that ratio different depending of speakers used?

[sportbilsentusiasten]

[s a minimum I recommend 75mm thick acoustic foam that is arranged on the wall, inward from the right loudspeaker (going towards the left loudspeaker), starting at the inside corner of the cabinet that is closest to the wall, and also starting at the bottom edge of the cabinet.

The panel should be about 60 x 60cm, extending about 25cm above the top of the cabinet.

Ideally, additional amounts and placement would be adapted to the needs of a particular environment.


For a more complete/optimal arrangement, as a general rule, I would suggest a 75mm thick, 150cm x 150cm square piece centered horizontally behind the cabinet, and starting at the floor.

Also, place 100mm thick acoustic foam outward from the speaker, (to the right of the right loudspeaker) over to the corner, and from the corner extend 1/3 the length of the sidewall, OR, a piece 1-meter wide centered on the point where the first specular reflection bounces off the sidewall to the listener.

Again, each room has it’s own issues to deal with, but these are the general guidelines for minimum and optimal absorption.

Let me know if I need to provide further explanation.

Great info,thanks!!!
What are your thoughts about acoustic treatment on the back wall, floor and ceiling?

[Jacro]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness? Why do I think carbon sounds more real than metal as a conductor? Any clue?

Objektivisten,

I was just curious... I have not seen any comment or feedback from you on the answers I provided so far to your questions.

It is a rather complex topic, so I thought I should check to see if my starting answers were useful for you, such that you would like me to continue, or not?

Best regards,

- James

Thanks for a very interesting an thoughtfull answer James. I have wait in silence and hope for the other topics, and I have read all your inputs in this thread and understand You have a had a lot to do. Please welcome.

Very good.

Thank you,

- James

[Jacro]

[s a minimum I recommend 75mm thick acoustic foam that is arranged on the wall, inward from the right loudspeaker (going towards the left loudspeaker), starting at the inside corner of the cabinet that is closest to the wall, and also starting at the bottom edge of the cabinet.

The panel should be about 60 x 60cm, extending about 25cm above the top of the cabinet.

Ideally, additional amounts and placement would be adapted to the needs of a particular environment.


For a more complete/optimal arrangement, as a general rule, I would suggest a 75mm thick, 150cm x 150cm square piece centered horizontally behind the cabinet, and starting at the floor.

Also, place 100mm thick acoustic foam outward from the speaker, (to the right of the right loudspeaker) over to the corner, and from the corner extend 1/3 the length of the sidewall, OR, a piece 1-meter wide centered on the point where the first specular reflection bounces off the sidewall to the listener.

Again, each room has it’s own issues to deal with, but these are the general guidelines for minimum and optimal absorption.

Let me know if I need to provide further explanation.

Great info,thanks!!!
What are your thoughts about acoustic treatment on the back wall, floor and ceiling?

Room treatments should be environment specific, but generally, I would recommend diffusion on the back-wall, and most everywhere else after the first specular reflections are absorbed.

At least first specular front-wall, floor, and ceiling reflections should be eliminated as much as possible, as they tend to be correlated and can cause the greatest sonic coloration and spatial compression distortion. After that, is good to keep the later reflections substantially live, but diffused to best attempt to simulate a larger, first venue type environment.

This is an over simplification on many counts. Bandwidth of the absorption is important as it is important not to just create a partially absorbed, spectrally imbalanced reflection contributing to the room sound.

- James

[2-ch]

Very interesting reading, i havent seen this thread earlier. I wonder how a guy from Seattle Washington found this place on the net?

Anyway, your are most welcome to faktiskt.se

[Jacro]

To achieve this, I believe the loudspeaker should either be operated as a boundary coupled half-space device, or as an omni or full range dipole well out into the room.

I agree on pretty much everything in this post and have a Q regarding this quote - please list a few of your favorite speakers during the years

Yes I read about your reluctance commenting on existing products so list discontinued speakers then.
(even if I consider giving positive feedback about a competing product something that gives cred )

Okay, here are a number of classic loudspeakers that I have either enjoyed and/or believe that they have contributed to advancing the state of the art. Most were very good performers with a few flaws, a few had significant flaws, but introduced important new capabilities or architectures to the field of loudspeaker development.
(I have not included any systems that I designed)

They are listed generally in chronological order of when they were produced.

Quad ESL 57
BBC LS3/6 – Spendor BC1
Hegeman H1 (Re-equalized)
Dayton Wright XG-8
Beveridge Model 2
Fulton FMI-80
BBC - Rogers LS3/5a (First build – Near-field or quasi-anechoic use)
Fulton J-Modular
Quad ESL-63 (modified and reoriented for optimized directivity)
AR-9
Snell Type ‘A’
Allison Model Three
Magneplanar Tympani 1D and Tympani IV
Acoustat 1+1
Snell Type One
Sound Labs A-1
3D Acoustics 3D6/10 (First version w/dust cap damping)
Carlsson OA Series
Gradient 1.3
Harbeth Monitor 40 (semi-near field)
AR-303

Here are a few currently available loudspeakers that, in my opinion, have promising characteristics.

AudioKinesis Dream Maker
Emerald Physics CS2.3
Ino Audio pi60
Gedlee Summa

Best regards,

- James

[Jacro]

Very interesting reading, i havent seen this thread earlier. I wonder how a guy from Seattle Washington found this place on the net?

Anyway, your are most welcome to faktiskt.se

I have written audio industry patent reviews for a loudspeaker engineering journal called "Voice Coil" for about 15 years. A few years ago, I was looking up some references on boundary coupled loudspeakers, for an article in the journal, and found Faktiskt.se for the first time when searching for additional information on Carlsson OA series and Audio Pro A4-14 loudspeakers.

I've always found it interesting to observe audio discussions from outside of the USA, and have particularly enjoyed Faktiskt.

I had not submitted any comments until just recently when a friend of mine told me that they noticed ( the clue ) loudspeaker being mentioned here. Since there appeared to be some questions and interest, as designer of the loudspeaker, I thought I should make myself available for discussion as long as there was an interest in my doing so.

I thank you, and everyone here, for warmly welcoming me into the clubhouse.

Cheers,

- James

[KarlXII]

Google translate is really a fantastic tool.

Jacro, have you listened to Carlsson and Ino yourself?


The Clue has really made people interested over here! Are there any plans to extend the series with e.g. a center speaker, surrounds or perhaps a big brother up front?

[Jacro]

Google translate is really a fantastic tool.

Jacro, have you listened to Carlsson and Ino yourself?


The Clue has really made people interested over here! Are there any plans to extend the series with e.g. a center speaker, surrounds or perhaps a big brother up front?

I hope that the Google translator doesn't provide too many embarrassing mistakes. I often translate from English to Swedish, and then translate the Swedish back to English again, to see what it says, and the errors can be quite entertaining.

Yes, I have heard the Carlsson and Ino loudspeakers. Sonab and Carlsson many years ago, and the larger Ino Audio loudspeakers more recently.

I have always been lucky enough to have customers and friends that have traveled all over the world, wanting to show me their latest loudspeaker discoveries that they bring back home to America.

I'm glad to hear that The Clue has generated interest here. I hope we can build enough to fill the U.S. orders so that eventually SJÖFN Hi-Fi can offer them in Europe. My understanding is that there have been orders from Europe but I don't know if they have been fulfilled.

In terms of future product plans for SJÖFN Hi-Fi, we will be focusing on support for The Clue and will be making some associated products to enhance the performance of The Clue.

As we announce any new products, I’ll post the announcement in Faktiskt.

Cheers,

- James

[paa]

James, I have just been reading a little of what Roy Allison have to say, and he apparently tries to get "maximum and equal dispersion
from the lowest to the highest audible frequencies" (in half space).
A tractrix or kugelwellenhorn and/or a small midrange with very wide dispersion, sounds like a possible way to achieve this in the upper range, but I cannot see how this goal could be compatible with using directivety control waveguides having the standard 60 to 90 degrees coverage angles?
Do you agree with Roy Allison that full half space is the optimum dispersion for the entire sound spectrum?

[Objektivisten]

Hello James.

[sportbilsentusiasten]

Thanks for the list Jacro!
I'm also a fan of QUAD, lovely speakers


Can you please elaborate on pi60's "promising characteristics"?

Here are a few currently available loudspeakers that, in my opinion, have promising characteristics.

AudioKinesis Dream Maker
Emerald Physics CS2.3
Ino Audio pi60

[Laila]

Kanske att man skall respektera vad Jacro tidigare
har skrivit, nämligen att han helst inte vill uttala sig om
typ "prylar" som nuvarande är i produktion, ity han
själv är i "branschen" . . .

[Jacro]

James, I have just been reading a little of what Roy Allison have to say, and he apparently tries to get "maximum and equal dispersion
from the lowest to the highest audible frequencies" (in half space).
A tractrix or kugelwellenhorn and/or a small midrange with very wide dispersion, sounds like a possible way to achieve this in the upper range, but I cannot see how this goal could be compatible with using directivety control waveguides having the standard 60 to 90 degrees coverage angles?
Do you agree with Roy Allison that full half space is the optimum dispersion for the entire sound spectrum?

paa,

It depends on the type of system architecture, boundary characteristics, and distance relationships among the closest four boundaries.

I tend to not agree with Allison, relative to the 180-degree radiation pattern being optimal, except for specialized applications, such as the Beveridge electrostatic or Bremen Egg when mounted on the sidewalls.

If I were NOT designing a dipole, I would generally prefer one of two approaches to the radiation pattern;

Either,

1) Limit a forward, monopole radiator to a maximum of 90-degrees over the full bandwidth (cross-firing in front of the listener), [[One of the ways this is accomplished through the use of a constant directivity, 90-degree waveguide combined with a proprietary type of gradient low frequency architecture]]

OR,

2) Arrange for the first four boundary distances to be cross-relational such that they are distributed and/or cross-cancelling. In the second case, the system would ideally, have a smooth, but complex transitional power response that is calibrated to match the predetermined boundary interaction.

Both are rather complex to develop, and the second approach is the basis for the operation of The Clue.

In terms of Allison’s systems, most all of them were 180-degree/half space horizontal radiators, and while they had a smooth balance and generous spaciousness, they were rather poor at focusing images, making the placement of each instrument more spatially vague than what the recording has to offer. Allison liked this type of spaciousness and vague imaging but it was not well received in America.

The exception was the less well-known Allison Model Three, which was meant to operate into a limited, 90-degree angle over the entire spectrum, and when properly set up with local absorption and an optimal room, was by far his best sounding product and an exceptional performer under ideal conditions.


- James

[petersteindl]

Kanske att man skall respektera vad Jacro tidigare
har skrivit, nämligen att han helst inte vill uttala sig om
typ "prylar" som nuvarande är i produktion, ity han
själv är i "branschen" . . .

Det existerar ingen INO högtalare. Det finns ingen INOhögtalare i produktion , eller?

Allt är ju på hobbybasis och i så fall existerar ingen produktion. Eller har jag förstått något felaktigt? Det är som jag ser det en fullkomlig omöjlighet att både ha en produktion och ha en produktion på hobbybasis.

MvH
Peter

[Objektivisten]

---

[paa]

... combined with a proprietary type of gradient low frequency architecture
...
- James

Thanks James for your interest in this forum and all the well written answers.

This proprietary gradient low frequency solution, is this something you have already used in any of your speaker designs?
Would it look something like this effort by Naqref on this forum:

Lilla Hjärtat - Historien om en falnad flamma..
http://www.faktiskt.se/modules.php?name ... 372#289372

The exception was the less well-known Allison Model Three, which was meant to operate into a limited, 90-degree angle over the entire spectrum, and when properly set up with local absorption and an optimal room, was by far his best sounding product and an exceptional performer under ideal conditions.
- James

But that is a speaker designed for corner placement, isn't it?
Then we have problems with driving all room modes to the max, and maybe a "hole in the middle", don't we?
How much wall absorbents would you put around corner speakers like these?

[JanBanan]

James,

You wrote in an earlier post that you have worked with Magnepan. May I ask what kind of input you provided and which model(s) you worked on?

[Jacro]

James,

You wrote in an earlier post that you have worked with Magnepan. May I ask what kind of input you provided and which model(s) you worked on?

Hi JanBanan,

My most recent work has been related to development of a new crossover topology that has been applied to the 1.7 and 3.7 models that were just introduced over the last 12 months or so.

I am basically a technical consultant that Magnepan brings into their development process to solve technical problems or provide training when their internal engineering group can’t find a solution or are having trouble optimizing particular design.

I am currently working on a new type of low frequency system.

They are very nice people and have been good friends since the mid-1970’s.

Best regards,

- James

[Jacro]

... combined with a proprietary type of gradient low frequency architecture
...
- James

Thanks James for your interest in this forum and all the well written answers.

This proprietary gradient low frequency solution, is this something you have already used in any of your speaker designs?
Would it look something like this effort by Naqref on this forum:

Lilla Hjärtat - Historien om en falnad flamma..
http://www.faktiskt.se/modules.php?name ... 372#289372

Hi paa,

No, the design that Naqref built is not the form of gradient loudspeaker that I was referring to, but I have developed the Naqref type a number of times, since the mid-1980’s (I refer to this special form of gradient as a “Di-Cardioid” [[Hybrid of Dipole and Cardioid]]) and was granted a patent (US 7,551,062) on a number of new variations of the concept.
(I also received a trademark on it: “SoundVector™”.)

I have used it in wide bandwidth loudspeakers I have experimented with over a couple decades, but about 5 years ago I developed a narrow-band, directional backup alarm for trucks, tractors, etc. for ECCO Safety Group, using this same basic architecture. If you are interested, here are a couple links on the project;

http://www.doriandrake.com/downloads/se ... 0Story.pdf

http://www.eccolink.com/Media/WhatsNews/SVX.pdf

This structure can be very effective, even more so if one makes an active version with two monopole transducers instead of one passive dipole.

I thought I was the first to invent it back in 1985, but it turns out that Bobby Beaver at Altec Lansing had developed products based on the concept at least 10 years earlier with the Model 814A “Extenda-Voice” and was granted a patent (US 3,722,616).

http://www.lansingheritage.org/html/alt ... rs/814.htm

Later work was done in 1982 by POLAR-PRO in Finland (WO8401681) and Skip Cross in America (US 4,437,541), and also in 2001 by Noselli in Italy, EP1137318A2.

Most recently, the technique can be seen in the Gradient Helsinki 1.5 loudspeaker, used as the midrange baffle:
http://www.gradient.fi/helsinki15/

When perfectly optimized, it can provide the best combination of efficiency, and angular cancellation of any first or second order gradient arrangement, clearly outperforming a simple cardioid (but not as effective as an end-fired array).

The gradient arrangement I referred to in my previous post is a unique arrangement that I have not seen used by anyone to date and accomplishes a null-zone in a novel manner that allows a better directivity match between a 90-degree waveguide and low frequency system crossing over to the waveguide. It is an approach that was created as a reference during the development of The Clue and was used in prototyping The Clue to create a tweeter gradient that cancels the tweeter energy on axis, so that it only adds off-axis energy to the larger woofer/midrange, or full range driver (up to a desired transition frequency).

I apologize if this is more information that desired, but this type of design has great potential that has not yet been realized in commercial products, so I thought that some in your group may enjoy seeing the reference materials for the architecture that Naqref (and Ingvar?) were experimenting with.
(By the way, the post of Naqref’s stopped after only 2 pages… Was there further work and measurements reported on the experimental devices?)

Best regards,

- James

[Jacro]

The exception was the less well-known Allison Model Three, which was meant to operate into a limited, 90-degree angle over the entire spectrum, and when properly set up with local absorption and an optimal room, was by far his best sounding product and an exceptional performer under ideal conditions.
- James

But that is a speaker designed for corner placement, isn't it?
Then we have problems with driving all room modes to the max, and maybe a "hole in the middle", don't we?
How much wall absorbents would you put around corner speakers like these?

Yes, the Allison Model Three is designed for corner placement.

You are correct that the corner placement will activate all the low frequency room modes. It is best used with at least two additional subwoofers with diversity placement to smooth the room response in the modal frequency range. Then the bass response is both strong and smooth.
(Room modes are fine if driven properly from at least 3 diverse locations).

Also, as you surmise, the “hole in the middle” problem can come into play if the room is too wide relative to the listening distance. That is a problem with corner placement… It only works in rooms that have the correct form factor that allows optimal listener/loudspeaker relationship.

With the correct room width-to-listener distance, there is no hole in the middle and the seating position, left to right, is less critical than with most systems.

The problematic reflections are very close the loudspeaker in this case, so damping material only needs to extend from the loudspeaker about a half-meter along the front and side walls.

I redesigned the Allison Three to have the midrange and tweeter placed rearward, much more closely coupled in the corner (the speaker face was about 35 cm wide at the bottom, and less than 10 cm wide at the top, near the tweeter), with a Snell type One ramps from the tweeter edge to the front and side walls.
With this arrangement one could operate with minimal damping material because of the absence of early reflections.

Very effective, if you have the ideal room layout.

- James

[Jacro]

Hell James

Objektivisten,

Curious... I'm not sure of what the significance of this statement is.

All the best,

- James

[petersteindl]

Hell James

Objektivisten,

Curious... I'm not sure of what the significance of this statement is.

All the best,

- James

There is an o missing.
It should have been Hello James Quite a big difference

Best Regards
Peter

[paa]

Hell James

Objektivisten,

Curious... I'm not sure of what the significance of this statement is.

All the best,

- James

If we assume it's correctly spelled, the swedish expression "hell" means "hail", although that word is not yet included in google translations.
Thanks for your answers again, by the way.

[Jacro]

Hell James

Objektivisten,

Curious... I'm not sure of what the significance of this statement is.

All the best,

- James

There is an o missing.
It should have been Hello James Quite a big difference

Best Regards
Peter

Thank you, Peter.

I feel better now.

- James

[Jacro]

Hell James

Objektivisten,

Curious... I'm not sure of what the significance of this statement is.

All the best,

- James

If we assume it's correctly spelled, the swedish expression "hell" means "hail", although that word is not yet included in google translations.
Thanks for your answers again, by the way.

paa,

Hell = Hail... That is good to know.
Thank you.

And, you are very welcome on the answers. I hope they are useful.

- James

[Objektivisten]

Holy macaroni, the significance of my 'statement' is zero (maybe below) James. Can we please go back to the real substance in this thread, your eminent inputs and answers. Regards

[JanBanan]

I am currently working on a new type of low frequency system.

Hell James,

This sounds exciting! Is this something you can elaborate on?

[KarlXII]

JanBanan, sometimes your particular sense of humour in combination with that avatar (which probably is a display of it) makes for a very ambiguous context.

[petersteindl]

Hell James

Objektivisten,

Curious... I'm not sure of what the significance of this statement is.

All the best,

- James

There is an o missing.
It should have been Hello James Quite a big difference

Best Regards
Peter

Thank you, Peter.

I feel better now.

- James

Very good

remember, we usually don't bite on this forum but sometimes we bark and sometimes we have a very weird humor not only janbanan.

I will introduce myself a little bit better later on. Take care.

Best regards
Peter

[Stereotypen]

Jacro:
It would be very interesting for me to hear what you think about the Audio Artistry Beethoven system. This is the speaker that I use myself. Here in Sweden there is a very limited amount of people that own this speaker but I guess that in the US they are more common. Have you heard them? What is your impression/oppinion of them? What is your oppinion of dipole bass in general?
The speakers are not manufactured any more so I guess it would be ok for you to comment on them.

[Jacro]

Jacro:
It would be very interesting for me to hear what you think about the Audio Artistry Beethoven system. This is the speaker that I use myself. Here in Sweden there is a very limited amount of people that own this speaker but I guess that in the US they are more common. Have you heard them? What is your impression/oppinion of them? What is your oppinion of dipole bass in general?
The speakers are not manufactured any more so I guess it would be ok for you to comment on them.

My apologies to everyone that has asked questions that I have not yet answered. I am particularly busy this week but will try to get back with more follow up next week.

I'll provide some short answers to a few questions and follow up later with more info.

Stereotypen,

Yes, I heard the Beethoven a few times back in 1998.

I thought that they could be made to perform quite well if they were in a large venue.

With the Linkwitz designs there are still some of the issues associated with center-symmetrical dipoles when they are constructed with point source drivers. This arrangement usually causes a significant dip in the midrange combined with some comb filtering ripple above the first dip frequency which is associated with the symmetrical 1-wavelength acoustical back-to-front path. This issue has been minimized with careful crossover work, but it is difficult to eliminate completely.

With dipoles, it is most often better to either use a full-baffle-sized diaphragm, such as with electrostatics or Magneplanars, OR, horizontally offset the transducer on the mounting baffle, to create diversity paths for the back-to-front output.

In terms of dipole bass, I believe it can be a very effective, at least when used above the lowest modal frequencies. Below the modal range, to maximize room gain, closed-form enclosures tend to be superior.

I've done quite a bit of work in this regard, in that I developed the first effective commercially viable cone based dipole woofer system, for use in the Carver Amazing Loudspeaker (with a Large Area Full Range Ribbon we developed for use above 100 Hz) in the early 1980's. Even though we were told it couldn't be done, we were able to achieve -3dB at 19 Hz by incorporating novel 12" woofers with 1.5" peak-to-peak linear excursion capability and a Qts of 3.4, mounted on a variable width baffle.

If anyone is interested, here is a picture of the system I developed...

http://www.soundsgoodtomehouston.com/carver/alspair.jpg

and

the European patent...

http://v3.espacenet.com/publicationDeta ... cale=en_EP

The folded version of a dipole woofer, such as used in the Beethovan subwoofer, can perform the bass function quite well, but there are some anomalies above the pass band that can make for more difficult crossover transitions to the upper range system than a simple flat/open dipole woofer system, but if miniaturization of the dipole is important, the folded type can reduce the apparent size effectively.

I don't mean to sound overly critical of the Beethoven. It is a system that has a lot of potential and I would very much like to see it developed further to reach that potential.

More later...

Cheers,

- James

[Stereotypen]

Thank you for your answer James!

[paa]

...
OR,

2) Arrange for the first four boundary distances to be cross-relational such that they are distributed and/or cross-cancelling. In the second case, the system would ideally, have a smooth, but complex transitional power response that is calibrated to match the predetermined boundary interaction.

Both are rather complex to develop, and the second approach is the basis for the operation of The Clue.
...
- James

How can this be made to work for different room sizes, and different listening distances, things that also would demand different distances between the speakers?
Could this cross-relational and cross-cancelling function be simulated in some known software?

[Naqref]

(By the way, the post of Naqref’s stopped after only 2 pages… Was there further work and measurements reported on the experimental devices?)

Hello James,

Finally I had som time (and focus) to read the whole thread and I'm very impressed as many others.

Regarding My little sweetheart it's scrapped but I'll think I have some measurement of it somewhere in one of my computers. Perhaps I'll find it and then I'll post some information here (and in the original thread).

The first ambience-module with the principle was made around 1995. A couple of years later I built the one on the pictures. I was still a poor student at the time so there was actually no money to develop the principle then. And then I found other things more promising.

I found your patent:
http://www.freepatentsonline.com/7551062.pdf
Impressive. I'll think the right way to go ist to use multiple active units just as you say. Passive damping of the sound from the back of the unit is to test intensive to get right.

I am like you intrested in what I call wall integrated speakers. It should perhaps be altered to external wall integrated speakers (EWI) or something like that to avoid beeing misstaken for in-wall-speakers. But anyhow. I'm the designer of Larsen speakers (it's a follow up of Carlsson). Two major problems with ewi-speakers is the perception of depth and excitment of roomresonances I'll think. The problem with depth is that the mind has problems to project auditory sources beyond the visual boundaries. The eye's override of other senses is making it harder to imagine a sound stage beyond the front wall. Do you think this is a big problem and if so what else is there to do about it other than removal of visual clues or to give additional visual clues to aid the sound stage?

The second problem (actually it's really a problem with almost every type of speaker), what is your preffered method of dealing with that? Acoustical lf-absorbers, parametrical eq or perhaps using multiple sound sources to counter act the resonanses?

Send my regards to Lars Erickson.

[paa]

I found the Bremen egg spekers to present a somewhat larger soundstage depthwise than sidewise when they were displayed at the Stockholm High End Show a couple of weeks ago. (With front speaker and proprietary matrix center decoding.)
Perhaps that was because they were placed on the sidewalls, positioned a bit back from the front wall, and the room was longer than wide, could that have been mainly a visible illusion too?

[petersteindl]

I found the Bremen egg spekers to present a somewhat larger soundstage depthwise than sidewise when they were displayed at the Stockholm High End Show a couple of weeks ago. (With front speaker and proprietary matrix center decoding.)
Perhaps that was because they were placed on the sidewalls, positioned a bit back from the front wall, and the room was longer than wide, could that have been mainly a visible illusion too?

That has primarily to do with the angle difference between the center and the side speakers. If I play the loudspeakers shifted so that the center is on the long wall and the Bremen eggspeakers on the short side walls, then it would have been different.

That was the way in which I originally intended to demonstrate my loudspeakers, but at the end just before moving into the room they said at Sheraton that the desk in the rooms could not be taken out and the desk is rather big, so I decided to do the demonstration as I did.

But the visual information is also important.

Best Regards
Peter Steindl

[Jacro]

Hi James, welcome to the forum, what do You think is the secret about tubes and that they almost always sounding more pleasant and true to the ears? And do we really need all that watts, low powered amps seems to gain in transparent sound, may be construction simplicity or cost effectiveness?

8O Holy macaroni, the significance of my 'statement' is zero (maybe below) James. Can we please go back to the real substance in this thread, your eminent inputs and answers. Regards

I agree.

In terms of our previous discussion on the causes of sonic differences between power amplifiers, in “Part One” discussed a few of the issues relating to the significance of overload characteristics during normal use of power amplifiers.

As a point of interest, these portions of the total research program that started in the 1970’s and were repeated in the early 1990’s as part of the development program for a new line of amplifiers that were produced in the mid and late 1990’s.

PART TWO: (A brief overview of the impact of wide band output impedance)

Next, in our research on this topic, we uncovered the effects of power amplifier output impedance and the interface with the loudspeaker load.

Output impedance is often referred to as damping factor relative to an 8-ohm load, but as we know, there are is not a single loudspeaker on the market that actually exhibits an 8-ohm, zero phase angle, load over the useful frequency range of the loudspeaker.

Also, damping factor is most often referred to as a low frequency phenomenon, with most amplifier specifications using 100 Hz as a reference frequency, when, in actuality, it is at least as important at high frequencies as it is at low frequencies. Unfortunately, most power amplifiers do not specify their high frequency damping factors… for the reason that most of them are substantially worse than they are at low frequencies.

(NOTE: There can also be dynamic, or non-linear output impedances in power amplifiers, and dynamic, non-linear input impedances in loudspeakers, but for this discussion, we will limit our observations to linear, small signal damping factor/output impedances and input impedances.)

Essentially, as one raises the output impedance (or lowers the damping factor) in a power amplifier, the amplitude error due to the impedance interface can cause the frequency response to take on the shape of the impedance curve more and more as the output impedance is higher.

Nearly all power amplifiers incorporate reactive components between the output stage and the output terminals. Most power amplifiers vary the amount of negative feedback vs. frequency, such that the output impedance increases with frequency due to the reduced feedback and output inductance.

After completing our studies in amplifier overload characteristics, we used the amplifiers in a manner that guaranteed that they operated below overload, and also kept them below the levels that would drive the loudspeaker in test, into nonlinearity, so as to eliminate those two variables in the device under test, as we moved on to isolate remaining parameters that affected sound quality in power amplifiers.

Even though we were operating both the power amplifiers and loudspeakers below and known nonlinear audible thresholds we were still detecting coloration differences between power amplifiers, with greater differences being exposed with some loudspeakers more than with other loudspeakers.

A first causal parameter, that we were able to isolate, was linear amplitude distortion due to the interaction of the amplifier and the loudspeaker impedances.

While there were audible errors in the lower and middle frequencies in certain tube amplifiers and a few specialized, low feedback solid-state amplifiers, (due to higher output impedances - poor damping factor - at low frequencies), this was not too surprising, as this issue had been reported previously, at least anecdotally, if not by way of rigorous controlled testing.

What was more interesting, were the number of amplifier loudspeaker combinations that generated high frequency coloration, due to the amplifier output impedance, at high frequencies, causing high frequency amplitude errors.

Many amplifiers have substantially worse damping factors at the highest frequencies, relative to the lower frequencies (below 1 kHz) and due to phase angle and impedance variations, exhibited in the loudspeaker, substantial response errors were observed. While most varied less than 1.2 dB, in some cases we found high-frequency response variations, between 5 kHz and 20 kHz, of over 2.5 dB!

As one can imagine, these levels of amplitude error, while in many cases subtle, are clearly audible, and repeatable with double blind testing.

It is interesting that there are still no industry standards for minimum impedance and phase angle. (THX and a few private parties have their own guidelines, but that is not recognized or enforced in most realms of audio).

It is easy for loudspeaker designers to be abusive in terms of impedance curves, as it can often serve their purposes to allow the impedance to fall to very low levels in exchange for some other advantage in performance, such as amplitude response.

Recently I was evaluating a new loudspeaker for a company, and I noticed that the impedance curve dipped to well below 2 Ohms at 285 Hz. They had done this because they had a dip in the amplitude response at 300 Hz and the only way to boost the response to flat in the passive-crossover, was to create an under-damped alignment in the high pass filter. The power spectrum in music program tends to be divided approximately equally above and below 270 Hz, and so the greater power demands are right were this loudspeaker had it’s lowest impedance, creating unusual current demands from the power amplifier.

They get the response they wanted and passed the impedance interface problem onto the amplifier designer.

Another popular loudspeaker, from the 1980’s, had a 0.9Ω impedance (and a large phase angle) at about 3.7 kHz. This caused a myriad of problems with amplifier interactions.

Amplifiers with any significant output impedance relative to the input impedance of the loudspeaker don’t operate any longer as an ideal voltage source, and can easily exhibit sonic errors because of this issue.

Depending on the set of conditions, audible problems can be caused by current limiting or by linear amplitude errors, or both.

As an aside, there is a story about our test procedure you may find interesting;

To isolate the audible effects during our testing process we used a number of approaches, such as straight wire bypass tests and double blind testing.

Some folks believe that results derived from blind testing is not valid, particularly when there is a null result.

During the time of our research, and still today, many audiophiles reject double blind testing as a procedure that causes stress that “interferes” with their psychological ability to make accurate assessments.

During our research project, we decided to test this notion. We had two amplifiers that both performed very well, but one sold for seven times more than the other one and was known as a very prestigious amplifier, having received rave reviews. We had two groups of audiophiles that claimed that they could always easily hear the difference between the two amplifiers, speaking rather critically of the lower cost unit, and glowingly of the higher cost amp.

During out double blind tests, each of them failed to consistently pick their favorite amplifier. So, we told them to take the two amplifiers, and listen to them under whatever test conditions they felt they were most comfortable, and to keep notes on what they perceived and which they preferred. We had approximately 20 people in two separate groups involved, all of them diehard audiophiles and critical of DBX testing.

Before we gave them the two amplifiers, we made one change. We exchanged the internal circuitry, so that the “cheap” amplifier was in the “expensive” chassis, and the “expensive” amp was in the “cheap” amp chassis.

We gave them the amplifiers and after they all had a chance to test them in the manner they were most comfortable, we retrieved the amplifiers and their written test results.

It may or may not be a surprise to you, that every one of them chose the amplifier in the expensive amplifier’s chassis as the one that sounded best, and had pages and pages of notes expressing their preference, and also expressing how easy it was to hear the difference and how the difference was quite large.

I learned two things from this experiment: 1) uncontrolled tests, that are not double blind, will produce random or erroneous results, and 2) even when shown this kind of evidence, it is not convincing for most of those that have already made up their mind that double blind testing doesn’t work.

So, I don’t think I made any converts, but it was an illuminating experiment.

Just to be clear, I am not saying that one cannot detect real differences between devices under test without a double blind test. I am only saying that it is important to verify any differences by way of double blind testing.

Actually, we have often been able to identify actual differences with double blind testing that are often not noticed in casual testing… but more on that another time.

Anyhow, so far we have covered two of the elements that can be of significance in explaining audible differences in power amplifiers, both of which are often overlooked;

- Audible clipping characteristics due to partial overload, and

- Wideband damping factor (Output Impedance) interface and interaction with the loudspeaker load

While there are some other issues that can affect amplifier sound quality, we can explain much of your original question with these two forms of distortion.

I can discuss further the specifics of these two forms and how they related to your original inquiry, and we can discuss other issues relative to power amplifiers in the future, if you wish, but I am delinquent on quite a few loudspeaker questions that I should answer first.

All the best,

- James

[Vinylcalle]

I have been reading this thread from the begining and want to say thanks to James for all the time he puts into giving us some great information. Especially the last answer about damping factor and amplifier research was great.

Well since I have got the opportunity I might as well ask you a question James.

Whats your view on vintage-hifi? I often get the feeling that there haven´t been much significant development since the golden days in the late 70s early 80s. If their is one thing that has changed its perhaps that there is more badly constructed equipment on the market today.
Some of my audiophile friends doesn´t agree, they´re basicly the ones who don´t believe in blind tests either.

Have you ever come across some of the old Denon totl equipment like poa3000,poa1500 and pma790? I think their great and some of the best amps I ever heard.

Best Regards Carl

[petersteindl]

Hi James, are you watching?

rgds
Peter

[Jacro]

Thanks for the list Jacro!
I'm also a fan of QUAD, lovely speakers


Can you please elaborate on pi60's "promising characteristics"?

Here are a few currently available loudspeakers that, in my opinion, have promising characteristics.

AudioKinesis Dream Maker
Emerald Physics CS2.3
Ino Audio pi60

The pi60 is a very organic and well integrated device with many good qualities.

But what impresses me most about the pi60, is not just its ability to perform well with excellent program material, but the unique capability to also extract a surprisingly lifelike presentation from many rather mediocre recordings.

That is not a trivial accomplishment.

- James

[Jacro]

Hi James, are you watching?

rgds
Peter

Yes, I am.

[Jacro]

I have been reading this thread from the begining and want to say thanks to James for all the time he puts into giving us some great information. Especially the last answer about damping factor and amplifier research was great.

Well since I have got the opportunity I might as well ask you a question James.

Whats your view on vintage-hifi? I often get the feeling that there haven´t been much significant development since the golden days in the late 70s early 80s. If their is one thing that has changed its perhaps that there is more badly constructed equipment on the market today.
Some of my audiophile friends doesn´t agree, they´re basicly the ones who don´t believe in blind tests either.

Have you ever come across some of the old Denon totl equipment like poa3000,poa1500 and pma790? I think their great and some of the best amps I ever heard.

Best Regards Carl

Hi Vinylcalle,

I am fond of many of the vintage units. The Denon POA-3000 was one of the better implementations of the Dynamic Class-A circuits that were being developed at that time (around 1979), and it was a beautiful package.

The PMA-790 Integrated utilized a variation on the Quad Current Dumping Architecture, that also worked quite well. I never had a chance to try the POA-1500.

I used to put on a vintage demonstration in Seattle every 5 years to show how well a pre-1970 system can be made to perform.

A double or triple set of Quad ESL's, with Marantz 9 power amplifiers, Marantz 7c pre-amp, and playing Master Tapes as source material was always good for illustrating how little progress has been made over the years.

- James

[paa]

Hello James,
Earl Geddes seems to say this is as good sound distribution as you need;



Could those curves be close to any goal you would want to reach?
Is it ok to have that much wider dispersion below 1 kHz than above?
Maybe the bass couples to the boundries and thus tends to narrow its dispersion a bit more than higher frequencies does?

[Jacro]

(By the way, the post of Naqref’s stopped after only 2 pages… Was there further work and measurements reported on the experimental devices?)

Hello James,

Finally I had som time (and focus) to read the whole thread and I'm very impressed as many others.

Regarding My little sweetheart it's scrapped but I'll think I have some measurement of it somewhere in one of my computers. Perhaps I'll find it and then I'll post some information here (and in the original thread).

The first ambience-module with the principle was made around 1995. A couple of years later I built the one on the pictures. I was still a poor student at the time so there was actually no money to develop the principle then. And then I found other things more promising.

I found your patent:
http://www.freepatentsonline.com/7551062.pdf
Impressive. I'll think the right way to go ist to use multiple active units just as you say. Passive damping of the sound from the back of the unit is to test intensive to get right.

I am like you intrested in what I call wall integrated speakers. It should perhaps be altered to external wall integrated speakers (EWI) or something like that to avoid beeing misstaken for in-wall-speakers. But anyhow. I'm the designer of Larsen speakers (it's a follow up of Carlsson). Two major problems with ewi-speakers is the perception of depth and excitment of roomresonances I'll think. The problem with depth is that the mind has problems to project auditory sources beyond the visual boundaries. The eye's override of other senses is making it harder to imagine a sound stage beyond the front wall. Do you think this is a big problem and if so what else is there to do about it other than removal of visual clues or to give additional visual clues to aid the sound stage?

The second problem (actually it's really a problem with almost every type of speaker), what is your preffered method of dealing with that? Acoustical lf-absorbers, parametrical eq or perhaps using multiple sound sources to counter act the resonanses?

Send my regards to Lars Erickson.

Hi Naqref,

Good to hear from you.

You mention two issues related to “EWI” loudspeakers:

(Over here we differentiate from "In-Wall" by calling them either; "On-Wall" or "Boundary Coupled"):

1) Perception of depth

2) Excitement of Room Resonances

I'm not clear on what you are referring to with #2. I think you are referring to reflections off the near boundary, but that is not actually a "resonance" per say. So, to discuss number 2, it would be good if you can clarify for me.

So, in terms of depth perception, in one way it is a problem (difficulty in imagining depth through a wall) and in another way, it is an improvement, in that it doesn't create "false depth" due to reflections off of the wall, spaced some distance behind the loudspeaker.

Loudspeakers spaced away from the front wall tend to create some degree of a false sense of depth depending on the ratio of the direct energy vs. the intensity of the energy reflected back to the listener.

So, in this regard, the boundary coupled systems are more faithful to reproducing the actual depth captured on the recording. (This is an over simplification, due to there being many additional variables determining perceived depth).

Your other observation, of the visual wall blocking our ability to imagine depth behind the wall, can be a significant limitation with some listeners. It is not a universal problem, but certainly a problem for many. Initially, I found it to be problematic for myself.

Sometimes I listen with my eyes closed, when listening to reproduced sound through loudspeakers, and also when I am at a live concert. In both cases, I find that the visual aspects alter my perception of the aural event, so closing my eyes creates a more equivalent situation for comparing live vs. reproduced sound.

If I have my eyes open at a live concert, I perceive the individual instruments as being more focused in a specific location in space than if my eyes are closed. Also, with eyes open, I "visualize" more depth of field, commensurate with the actual stage depth. If I close my eyes at a live concert, then the location of instruments is more vague, less focused, and the perception of depth of field is less.

So, if we are attempting to transport the actual focus, and scale of the sound sources to our listening room, without the actual musicians being visually present, the boundary coupled system is actually a more “accurate” facsimile of the spatial dimension of the original event.

(In a smaller second venue, boundary coupled loudspeakers and loudspeakers out in the room both distort the image compared to the original, but the loudspeakers that are out in the room (unless highly directional) will add more artificiality to the reproduced sound.

To minimize the visual barrier (for some customers that find it bothersome), but still have a boundary-coupled relationship for the loudspeakers, I have developed an alternative use-model.

Let’s see if I can explain it without a drawing.

Instead of facing the front wall of the room, the listener faces into a corner of the room. As an example, the listener might sit 3-meter’s straight out from the corner, and the left loudspeaker is coupled to the wall that extends to the left of the corner, and the other loudspeaker is coupled to the wall that extends to the right of the corner. The distance between the two loudspeakers and the distance from the loudspeakers to the listener are the same as they normally would be, and the loudspeakers are angled, or aimed, in relation to the listener, as they normally would be.

Visually, the corner extends back behind the loudspeakers, but the loudspeakers are effectively boundary coupled to each wall as they normally would be on the front wall.

I usually put extra absorption/diffusion in the corner, such as a quarter round of a half-meter radius, covered with 100 mm of foam, and I use a larger piece of absorption on the wall where the loudspeaker is coupled.

With an angled configuration, such as the Larsen 4, 6, or 8, the right loudspeaker may work better on the left side, and visa versa.

Make sense?

If properly set up, this has a few advantages, two of which are; it provides a visual expansion of depth, and it can leverage a more effective boundary coupling (less than a half-space) which allows a small loudspeaker to better power a larger room.

I use this approach for listeners that find the wall boundary mounting to be problematic for imagining depth, and/or for listeners who have rooms that are larger than what the loudspeaker is normally capable of filling.

This is a rather unorganized, non-technical discussion of the issue that you raised, but hopefully it is still somewhat useful.

I’ll address the rest later.

Best regards,

- James

PS – I’ll be sure to say hello to Lars for you.

[Jacro]

Hello James,
Earl Geddes seems to say this is as good sound distribution as you need;



Could those curves be close to any goal you would want to reach?
Is it ok to have that much wider dispersion below 1 kHz than above?
Maybe the bass couples to the boundries and thus tends to narrow its dispersion a bit more than higher frequencies does?

paa,

I have to run to a meeting, so a short answer for now.

While I believe that Geddes radiation pattern of constant directivity above 1 kHz is a big improvement over most free-standing loudspeakers, I disagree with Geddes suggesting that it is not important to maintain the constant radiation pattern below 1 kHz. A 12 dB transition below 1.5 kHz still impacts the “room sound”.

Geddes admits that he is not sure that it is important to improve it further, but states that, “even if it is important, it is not practical to do so”.

I humbly disagree.

But, I must also say, that Geddes has done some excellent work, his approach being superior to 99% of the loudspeakers available today, and he is one of the few actually advancing the art.

So, I agree that his configuration is an improvement over most of the other approaches, I merely feel that he has stopped short of what can be achieved on an ideal basis… definitely with his actual device, and possibly with his belief.

He is on the right track… he just needs to take it a bit further.

More later…

- James

[Jacro]

I am currently working on a new type of low frequency system.

Hell James,

This sounds exciting! Is this something you can elaborate on?

JanBanan,

I'm sorry, but that is confidential at this time.

If and when it can be disclosed publicly, I will be glad to discuss it here.

- James

[paa]

Thanks James for your answer about directivety below 1kHz.
Now imagine that I manage to make loudspeakers that maintain directivety down to say 80Hz, and want to add some subwoofers to them:
If I place two subs in the front corners, they will be boundry coupled in their frequency range and have 90+90 degrees of dispersion which should be fine. But probably we play the subs in mono so now we have 180 degrees bass dispersion which is quite a bit more more than the theoretically perfect main speakers, how do we sort this out? And if we want one or two more subs to improve on room modes, how would all that effect frequency response vs power response in such a system?

[LypsylateX]

I hope that the Google translator doesn't provide too many embarrassing mistakes. I often translate from English to Swedish, and then translate the Swedish back to English again, to see what it says, and the errors can be quite entertaining.

It helps that these errors is mostly one-way, i.e. when you're reading swedish threads. I'd say that 75% (just a guess) around here have no problems using english as primary language on a forum, and half of those should have no problem with quite advanced technical english.

Not a single lecture in my university education, in sweden, were in swedish. It was all in english. That helps. I guess this goes for most of the other people here with any form of higher technical education aswell. For me it's not the language barrier that sometimes keeps me from understanding threads like these, it's lack of physics knowledge.

My beleif is that this kind of participation from you could be alot more difficult in some other languages. (Like... well... french for instance. That would be very difficult for the majority of us swedes aswell, for the same reason.)

[Naqref]

(By the way, the post of Naqref’s stopped after only 2 pages… Was there further work and measurements reported on the experimental devices?)

Hello James,

Finally I had som time (and focus) to read the whole thread and I'm very impressed as many others.

Regarding My little sweetheart it's scrapped but I'll think I have some measurement of it somewhere in one of my computers. Perhaps I'll find it and then I'll post some information here (and in the original thread).

The first ambience-module with the principle was made around 1995. A couple of years later I built the one on the pictures. I was still a poor student at the time so there was actually no money to develop the principle then. And then I found other things more promising.

I found your patent:
http://www.freepatentsonline.com/7551062.pdf
Impressive. I'll think the right way to go ist to use multiple active units just as you say. Passive damping of the sound from the back of the unit is to test intensive to get right.

I am like you intrested in what I call wall integrated speakers. It should perhaps be altered to external wall integrated speakers (EWI) or something like that to avoid beeing misstaken for in-wall-speakers. But anyhow. I'm the designer of Larsen speakers (it's a follow up of Carlsson). Two major problems with ewi-speakers is the perception of depth and excitment of roomresonances I'll think. The problem with depth is that the mind has problems to project auditory sources beyond the visual boundaries. The eye's override of other senses is making it harder to imagine a sound stage beyond the front wall. Do you think this is a big problem and if so what else is there to do about it other than removal of visual clues or to give additional visual clues to aid the sound stage?

The second problem (actually it's really a problem with almost every type of speaker), what is your preffered method of dealing with that? Acoustical lf-absorbers, parametrical eq or perhaps using multiple sound sources to counter act the resonanses?

Send my regards to Lars Erickson.

Hi Naqref,

Good to hear from you.

You mention two issues related to “EWI” loudspeakers:

(Over here we differentiate from "In-Wall" by calling them either; "On-Wall" or "Boundary Coupled"):

1) Perception of depth

2) Excitement of Room Resonances

I'm not clear on what you are referring to with #2. I think you are referring to reflections off the near boundary, but that is not actually a "resonance" per say. So, to discuss number 2, it would be good if you can clarify for me.

So, in terms of depth perception, in one way it is a problem (difficulty in imagining depth through a wall) and in another way, it is an improvement, in that it doesn't create "false depth" due to reflections off of the wall, spaced some distance behind the loudspeaker.

Loudspeakers spaced away from the front wall tend to create some degree of a false sense of depth depending on the ratio of the direct energy vs. the intensity of the energy reflected back to the listener.

So, in this regard, the boundary coupled systems are more faithful to reproducing the actual depth captured on the recording. (This is an over simplification, due to there being many additional variables determining perceived depth).

Your other observation, of the visual wall blocking our ability to imagine depth behind the wall, can be a significant limitation with some listeners. It is not a universal problem, but certainly a problem for many. Initially, I found it to be problematic for myself.

Sometimes I listen with my eyes closed, when listening to reproduced sound through loudspeakers, and also when I am at a live concert. In both cases, I find that the visual aspects alter my perception of the aural event, so closing my eyes creates a more equivalent situation for comparing live vs. reproduced sound.

If I have my eyes open at a live concert, I perceive the individual instruments as being more focused in a specific location in space than if my eyes are closed. Also, with eyes open, I "visualize" more depth of field, commensurate with the actual stage depth. If I close my eyes at a live concert, then the location of instruments is more vague, less focused, and the perception of depth of field is less.

So, if we are attempting to transport the actual focus, and scale of the sound sources to our listening room, without the actual musicians being visually present, the boundary coupled system is actually a more “accurate” facsimile of the spatial dimension of the original event.

(In a smaller second venue, boundary coupled loudspeakers and loudspeakers out in the room both distort the image compared to the original, but the loudspeakers that are out in the room (unless highly directional) will add more artificiality to the reproduced sound.

To minimize the visual barrier (for some customers that find it bothersome), but still have a boundary-coupled relationship for the loudspeakers, I have developed an alternative use-model.

Let’s see if I can explain it without a drawing.

Instead of facing the front wall of the room, the listener faces into a corner of the room. As an example, the listener might sit 3-meter’s straight out from the corner, and the left loudspeaker is coupled to the wall that extends to the left of the corner, and the other loudspeaker is coupled to the wall that extends to the right of the corner. The distance between the two loudspeakers and the distance from the loudspeakers to the listener are the same as they normally would be, and the loudspeakers are angled, or aimed, in relation to the listener, as they normally would be.

Visually, the corner extends back behind the loudspeakers, but the loudspeakers are effectively boundary coupled to each wall as they normally would be on the front wall.

I usually put extra absorption/diffusion in the corner, such as a quarter round of a half-meter radius, covered with 100 mm of foam, and I use a larger piece of absorption on the wall where the loudspeaker is coupled.

With an angled configuration, such as the Larsen 4, 6, or 8, the right loudspeaker may work better on the left side, and visa versa.

Make sense?

If properly set up, this has a few advantages, two of which are; it provides a visual expansion of depth, and it can leverage a more effective boundary coupling (less than a half-space) which allows a small loudspeaker to better power a larger room.

I use this approach for listeners that find the wall boundary mounting to be problematic for imagining depth, and/or for listeners who have rooms that are larger than what the loudspeaker is normally capable of filling.

This is a rather unorganized, non-technical discussion of the issue that you raised, but hopefully it is still somewhat useful.

I’ll address the rest later.

Best regards,

- James

PS – I’ll be sure to say hello to Lars for you.

What you say makes a lot of sense. I hade tried to convince people of using the adjacent walls in a corner and shift right and left speaker in some cases but the look at me like they think; "Are you crazy or something" Perhaps I should work on my way of convincing people more to adress this problem.

Anyhow I think the problem is non existing when you have a picture of some sort between the boundary coupled (better word than mine at least in English) speakers. Like a TV or a projector screen.
Perhaps using mirrors (with some parts of the surface that are opaque) could do the trick?


My second question was about the normal resonances you have in a rectangular room (where you always have pressure nodes and velocity antinodes in the corners). A boundary coupled (just because it's places close to the walls) excite or trigger thes resonances more than free-standing speakers. At higher frequencys as you know the resonances have a modal density that is so high that single resonanses can't be heard. But at low frequencys (below 100 Hz or so) they can be quite problematic. As I see it there are (at least) three methods that can be used against these resonanses;
1. Using absorbtion (helmholtz, mass surface absorbers or large normal sound absorbers).
2. Using multiple or distributed sound sources. If you place a single sound source at in a place in the room where there is a pressure node for a certain frequency then it will trigger that corresponding resonance. But if you place a second correlated sound source at another pressure node for that frequency then that resonance can't be triggered. One way to use this technique is to use several subwoofers for the range of 0-100Hz and place them in a pattern that supresses these resonances.
3. Using a parametric eq.

Does my explanation makes my question more clear?

Reagards
Anders Eriksson

[Jacro]

Thanks James for your answer about directivety below 1kHz.
Now imagine that I manage to make loudspeakers that maintain directivety down to say 80Hz, and want to add some subwoofers to them:
If I place two subs in the front corners, they will be boundry coupled in their frequency range and have 90+90 degrees of dispersion which should be fine. But probably we play the subs in mono so now we have 180 degrees bass dispersion which is quite a bit more more than the theoretically perfect main speakers, how do we sort this out? And if we want one or two more subs to improve on room modes, how would all that effect frequency response vs power response in such a system?

paa,

My computer is in for repair for a few days, so I will have to wait to address this question more completely as soon as my computer is returned to me.

You ask a good question. Part of the answer is that it is most important to maintain the directivity down to the Schroeder frequency. Below the Schroeder frequency things change and to some degree, the sound source becomes the room modes themselves, as opposed to the loudspeaker, but that is a concept that needs some further explanation.

I hope to get back to you with more on this topic in a few days.

All the best,

- James

[Jacro]

(By the way, the post of Naqref’s stopped after only 2 pages… Was there further work and measurements reported on the experimental devices?)

Hello James,

Finally I had som time (and focus) to read the whole thread and I'm very impressed as many others.

Regarding My little sweetheart it's scrapped but I'll think I have some measurement of it somewhere in one of my computers. Perhaps I'll find it and then I'll post some information here (and in the original thread).

The first ambience-module with the principle was made around 1995. A couple of years later I built the one on the pictures. I was still a poor student at the time so there was actually no money to develop the principle then. And then I found other things more promising.

I found your patent:
http://www.freepatentsonline.com/7551062.pdf
Impressive. I'll think the right way to go ist to use multiple active units just as you say. Passive damping of the sound from the back of the unit is to test intensive to get right.

I am like you intrested in what I call wall integrated speakers. It should perhaps be altered to external wall integrated speakers (EWI) or something like that to avoid beeing misstaken for in-wall-speakers. But anyhow. I'm the designer of Larsen speakers (it's a follow up of Carlsson). Two major problems with ewi-speakers is the perception of depth and excitment of roomresonances I'll think. The problem with depth is that the mind has problems to project auditory sources beyond the visual boundaries. The eye's override of other senses is making it harder to imagine a sound stage beyond the front wall. Do you think this is a big problem and if so what else is there to do about it other than removal of visual clues or to give additional visual clues to aid the sound stage?

The second problem (actually it's really a problem with almost every type of speaker), what is your preffered method of dealing with that? Acoustical lf-absorbers, parametrical eq or perhaps using multiple sound sources to counter act the resonanses?

Send my regards to Lars Erickson.

Hi Naqref,

Good to hear from you.

You mention two issues related to “EWI” loudspeakers:

(Over here we differentiate from "In-Wall" by calling them either; "On-Wall" or "Boundary Coupled"):

1) Perception of depth

2) Excitement of Room Resonances

I'm not clear on what you are referring to with #2. I think you are referring to reflections off the near boundary, but that is not actually a "resonance" per say. So, to discuss number 2, it would be good if you can clarify for me.

So, in terms of depth perception, in one way it is a problem (difficulty in imagining depth through a wall) and in another way, it is an improvement, in that it doesn't create "false depth" due to reflections off of the wall, spaced some distance behind the loudspeaker.

Loudspeakers spaced away from the front wall tend to create some degree of a false sense of depth depending on the ratio of the direct energy vs. the intensity of the energy reflected back to the listener.

So, in this regard, the boundary coupled systems are more faithful to reproducing the actual depth captured on the recording. (This is an over simplification, due to there being many additional variables determining perceived depth).

Your other observation, of the visual wall blocking our ability to imagine depth behind the wall, can be a significant limitation with some listeners. It is not a universal problem, but certainly a problem for many. Initially, I found it to be problematic for myself.

Sometimes I listen with my eyes closed, when listening to reproduced sound through loudspeakers, and also when I am at a live concert. In both cases, I find that the visual aspects alter my perception of the aural event, so closing my eyes creates a more equivalent situation for comparing live vs. reproduced sound.

If I have my eyes open at a live concert, I perceive the individual instruments as being more focused in a specific location in space than if my eyes are closed. Also, with eyes open, I "visualize" more depth of field, commensurate with the actual stage depth. If I close my eyes at a live concert, then the location of instruments is more vague, less focused, and the perception of depth of field is less.

So, if we are attempting to transport the actual focus, and scale of the sound sources to our listening room, without the actual musicians being visually present, the boundary coupled system is actually a more “accurate” facsimile of the spatial dimension of the original event.

(In a smaller second venue, boundary coupled loudspeakers and loudspeakers out in the room both distort the image compared to the original, but the loudspeakers that are out in the room (unless highly directional) will add more artificiality to the reproduced sound.

To minimize the visual barrier (for some customers that find it bothersome), but still have a boundary-coupled relationship for the loudspeakers, I have developed an alternative use-model.

Let’s see if I can explain it without a drawing.

Instead of facing the front wall of the room, the listener faces into a corner of the room. As an example, the listener might sit 3-meter’s straight out from the corner, and the left loudspeaker is coupled to the wall that extends to the left of the corner, and the other loudspeaker is coupled to the wall that extends to the right of the corner. The distance between the two loudspeakers and the distance from the loudspeakers to the listener are the same as they normally would be, and the loudspeakers are angled, or aimed, in relation to the listener, as they normally would be.

Visually, the corner extends back behind the loudspeakers, but the loudspeakers are effectively boundary coupled to each wall as they normally would be on the front wall.

I usually put extra absorption/diffusion in the corner, such as a quarter round of a half-meter radius, covered with 100 mm of foam, and I use a larger piece of absorption on the wall where the loudspeaker is coupled.

With an angled configuration, such as the Larsen 4, 6, or 8, the right loudspeaker may work better on the left side, and visa versa.

Make sense?

If properly set up, this has a few advantages, two of which are; it provides a visual expansion of depth, and it can leverage a more effective boundary coupling (less than a half-space) which allows a small loudspeaker to better power a larger room.

I use this approach for listeners that find the wall boundary mounting to be problematic for imagining depth, and/or for listeners who have rooms that are larger than what the loudspeaker is normally capable of filling.

This is a rather unorganized, non-technical discussion of the issue that you raised, but hopefully it is still somewhat useful.

I’ll address the rest later.

Best regards,

- James

PS – I’ll be sure to say hello to Lars for you.

What you say makes a lot of sense. I hade tried to convince people of using the adjacent walls in a corner and shift right and left speaker in some cases but the look at me like they think; "Are you crazy or something" Perhaps I should work on my way of convincing people more to adress this problem.

Anyhow I think the problem is non existing when you have a picture of some sort between the boundary coupled (better word than mine at least in English) speakers. Like a TV or a projector screen.
Perhaps using mirrors (with some parts of the surface that are opaque) could do the trick?


My second question was about the normal resonances you have in a rectangular room (where you always have pressure nodes and velocity antinodes in the corners). A boundary coupled (just because it's places close to the walls) excite or trigger thes resonances more than free-standing speakers. At higher frequencys as you know the resonances have a modal density that is so high that single resonanses can't be heard. But at low frequencys (below 100 Hz or so) they can be quite problematic. As I see it there are (at least) three methods that can be used against these resonanses;
1. Using absorbtion (helmholtz, mass surface absorbers or large normal sound absorbers).
2. Using multiple or distributed sound sources. If you place a single sound source at in a place in the room where there is a pressure node for a certain frequency then it will trigger that corresponding resonance. But if you place a second correlated sound source at another pressure node for that frequency then that resonance can't be triggered. One way to use this technique is to use several subwoofers for the range of 0-100Hz and place them in a pattern that supresses these resonances.
3. Using a parametric eq.

Does my explanation makes my question more clear?

Reagards
Anders Eriksson

Anders,

As I mentioned in the last post, I am without my computer this week, so I will have to get back to you with a complete answer in a few days.

Yes, I do understand your question now... thank you for clarifying.

There are a number of issues around the topic that you have raised.

The solutions that you mention, or variations on them, tend to be some of the best we have available today.

To make a couple statements until I can get back with complete answers, I would say that I am generally an advocate of "Global", preemptive, solutions (such as modifying the room, loudspeaker configuration and location, and using multiple woofer systems), as opposed to "local", post-corrective solutions (such as digital room correction).

There are a number of ways to implement multiple woofers. I have a new couple of new ways that I have found that exhibit interesting results when compared to some of the current conventional approaches of Welti and Geddes. I'll speak more about them when I get back to complete my answers.

More later...

Best regards,

- James

[Jacro]

What is according to you the best length to width ratio?
Is that ratio different depending of speakers used?

Hi sportbilsentusiasten,

My computer is repaired and I am back online. I apologize for the delay.

I have many questions to respond to here.

Relative to your question, the most effective, angle and length-to-width ratio depends on system type and program material. While there are a numerous optimal angles, due to the wide variety of system configurations, I will just list three useful angles that tend to maximize the performance of many of the most common program sources and loudspeaker types.

1) Standard Stereo:
For a simple two-channel system with standard stereo program material, plus and minus 21 degree angles for the left and right loudspeakers tend to work best.

This is not so much due to recording type, which may vary quite a bit based on microphone usage. Instead, the angle is defined placement that provides the least amount of inter-aural crosstalk. This form of “passive” crosstalk cancellation partially mimics binaural or crosstalk cancellation systems. By doing so, it can recreate the perception of the most effective three-dimensional sound field, with “balanced” depth and spatial characteristics.

2) Three Channel:
For systems with a center channel loudspeaker, the purpose is to anchor the centermost information, with low coloration and spatial correctness in multiple listening seats (unfortunately it often introduces a problem of monophonic scale on center channel signals) but with this approach there are some spatial errors that force a different placement for the left and right loudspeaker if one wishes to maintain the full dimensionality of the sound field. One of the reasons is that a hard left signal is normally heard as placed at the left loudspeaker, in a two-channel arrangement. But, with a center channel system, the hard left signal will be placed between the center and left loudspeaker as a phantom image. If the left and right loudspeakers are kept at the same 21-degree angles, the stage width will be limited to about +/- 10.5 degrees.

So, with the center channel addition, we need to change the angular staging of the left and right channel loudspeakers. It turns out that an angle of approximately +/- 67.5-degrees is required to sustain the full stage width when using a center channel device. This wide of an optimal angle is due to a number of issues, including the fact the image spread now being limited by the phantom source width limit, and, also because the speakers can no longer be positioned at the 21-degree inter-aural cancellation angle and therefore they loose the ability enhance spaciousness due to a lack of optimal IAC.

3)Two channel program material that is Cross talk cancellation processed:
As a third angular specification, is that of optimizing for recordings that have been programmed with spatial enhancing crosstalk cancellation or are being post processed with a crosstalk cancellation system. For these recordings it is recommended that the loudspeakers place with angles that are less than plus/minus 15 degrees, and preferably on the order of plus/minus 5 degrees. This is implemented partially to minimize the variation of crosstalk cancellation and head related amplitude response vs. frequency.

What this means is that ideally, one moves their loudspeakers to have different angles relative to the listener depending on the nature of the program source.

As with most all of the issues that we are discussing, this discussion is not at all comprehensive due to both the wide range of program material, channel count and type, and also the myriad of other factors (frequency response, room interaction, etc.) that are altered as one changes the relationship of the two loudspeakers to each other and the listener.

****
To everyone:
I have lost track of which issues remain unanswered. If there are any questions that someone has submitted for me to answer that I have not responded to, or if there any that I have not answered completely of which there is still interest in further discussion, please let me know.

All the best,

- James

[paa]

I have lost track of which issues remain unanswered. If there are any questions that someone has submitted for me to answer that I have not responded to, or if there any that I have not answered completely of which there is still interest in further discussion, please let me know.?

I believe this is long lost:

...
OR,

2) Arrange for the first four boundary distances to be cross-relational such that they are distributed and/or cross-cancelling. In the second case, the system would ideally, have a smooth, but complex transitional power response that is calibrated to match the predetermined boundary interaction.

Both are rather complex to develop, and the second approach is the basis for the operation of The Clue.
...
- James

How can this be made to work for different room sizes, and different listening distances, things that also would demand different distances between the speakers?
Could this cross-relational and cross-cancelling function be simulated in some known software?

Then there was the discussion about power response with multiple subwoofers, where you said you had something more coming.

[Harryup]

To make a couple statements until I can get back with complete answers, I would say that I am generally an advocate of "Global", preemptive, solutions (such as modifying the room, loudspeaker configuration and location, and using multiple woofer systems), as opposed to "local", post-corrective solutions (such as digital room correction).

There are a number of ways to implement multiple woofers. I have a new couple of new ways that I have found that exhibit interesting results when compared to some of the current conventional approaches of Welti and Geddes. I'll speak more about them when I get back to complete my answers.

More later...

Best regards,

- James

Jacro,
I have followed the thread with great interest and you are really a good asset to have here.
I have made some test with multiple subwoofers and active digital crossovers (BSS FDS-366T) and I have found rather huge aural differences when using the time delay function. Do you recommend using multiple subbs without delay adjustments? From my small test with multiple subbs using Smaart I found that adjusting the delay for a rather straight frequency response will give a better transient response overall compared to rather straight frequency response using the peq. Are my observations correlated to using an active digital crossover which has an already built-in delay of at least 2ms? Could it be the case that just using an analog active crossover and using several subbs will "cure" all timing problems?
Unfortunately I have not the possibility to test multiple subbs with analog crossover except at home where I just have 2 Paradigm Servo-15 directly placed under the midrange and treble speakers. There I have compared an analogue JBL553 with the BSS and if I sit just a bit off center I could never prefer the analogue time uncompensated filter. In the middle of the sweet spot the difference is less but a dentist chair is preferable.
What is the risk of getting a rather straight frequency response using multiple subbs and losing some transient response due to have the subbs playing a bit in and out of phase with each other and the rest of the system?
I have looked into jevans111 small movies at youtube
http://www.youtube.com/results?search_query=subwoofer+alignment&aq=f about subwoofer alignment and I'm curious how to find out when the setup is fully optimized. What tool would you recommend or is it even a correct approach for home environments.

Regards
Harryup

[avr7000]

Hello Jacro

Is there a possibly somewhere for me to listen to, and maybe buy, the clue speakers in either:
New York
Grand Rapids
Richmond
-and maybe Chicago.

Regards
Stefan

[Jacro]

Hello Jacro

Is there a possibly somewhere for me to listen to, and maybe buy, the clue speakers in either:
New York
Grand Rapids
Richmond
-and maybe Chicago.

Regards
Stefan

Hi Stefan,

We have distribution that is ready to be set up in New York as soon as we are fully able to supply our other distributors, hopefully in the next two to three months.

We have nothing available in Grand Rapids or Richmond at this time.

Currently, we DO have an outlet in Chicago. The contact is Paul Lawless ( paul@sjofnhifi.com )

I hope this can work for you. If not, let me know and I'll alert you when we activate our distribution in New York.

Best regards,

- James

[Jacro]

I have lost track of which issues remain unanswered. If there are any questions that someone has submitted for me to answer that I have not responded to, or if there any that I have not answered completely of which there is still interest in further discussion, please let me know.?

I believe this is long lost:

...
OR,

2) Arrange for the first four boundary distances to be cross-relational such that they are distributed and/or cross-cancelling. In the second case, the system would ideally, have a smooth, but complex transitional power response that is calibrated to match the predetermined boundary interaction.

Both are rather complex to develop, and the second approach is the basis for the operation of The Clue.
...
- James

How can this be made to work for different room sizes, and different listening distances, things that also would demand different distances between the speakers?
Could this cross-relational and cross-cancelling function be simulated in some known software?

Then there was the discussion about power response with multiple subwoofers, where you said you had something more coming.

paa,

I can’t go into all the subtleties and complexity of applying the cross-relational techniques without applying at least ten pages of writing, but I’ll state a few useful starting points to allow one to engage the general concept and get somewhat of a feel for variation relating to room dimension differences.

If the loudspeaker low frequency system is calibrated to the appropriate alignment with the correct high pass slope, and the placement of transducers and cabinet dimensions are properly integrated with the total scheme, then the desirable impact of the upper-most sensitive boundary frequencies (less than 0.75 wL) will be dominated from correct distances being established to the floor and front wall. The secondary relationships will be the sidewalls, ceiling and mutual coupling interactions. This relationship has implications for both the low frequency extension, AND, the smoothness of the response due to direct and the multiple reflected sounds being coordinated for diverse, distributed relationships. Without resorting to the optimal complexity, one approach that can work well is to apply “golden rule” relationships (1.618 x) between each of the specular sources.

The secondary sources have the greatest impacts below the Schroeder frequency, in the modal region. While the optimization of these dimensions are important (and may be interchangeable or reordered, if one of them must change outside an optimum range) have a certain amount of inherent flexibility due to modal dominance.

Yes, the optimization can be simulated and arrived at by way of software, and is currently the subject of a patent application.

There is much more to these relationships, but I would add that a two-loudspeaker system, in most cases, will not provide smooth response in the modal region. The application of multiple diversity woofers will most often be required to optimize the room/system response.

These are complex subjects, the room relationship of the loudspeaker, and also the application of multiple subwoofers.

There are a number of complex, simulation and measurement-based calibration of multiple subwoofers, but a few years ago I developed a simplified system for my customers to effectively apply multiple subwoofers to achieve excellent results with minimal analysis.

It is based on the use of a subwoofer on each of the four walls, with the relationship of each woofer being that of an increasing 1.618 relationship of each woofer to a corner. The first woofer is placed at a distance from a first corner that is based on a specific value relating to a constant relative to room dimensions. The second woofer is placed 1.618 times that distance from the second corner. Third woofer placed, relative to the third corner, 1.618 times the distance of the second woofer, and the fourth, 1.618 of the third. The phase of woofers 2, 3, and 4 are tested individually with reversed phase (usually woofer 2 or 3 work best with phase reversed)
This simple approach produces surprisingly consistent results.

Okay, back to work… I will try to write more later on system power response issues above and below the Schroeder frequency.

Cheers,

- James

[paa]

Hi James,
Once again, thanks for your patience with all our questions.
However much I await more about these things I have a follow up question here.

...
It is based on the use of a subwoofer on each of the four walls, with the relationship of each woofer being that of an increasing 1.618 relationship of each woofer to a corner. The first woofer is placed at a distance from a first corner that is based on a specific value relating to a constant relative to room dimensions. The second woofer is placed 1.618 times that distance from the second corner. Third woofer placed, relative to the third corner, 1.618 times the distance of the second woofer, and the fourth, 1.618 of the third. The phase of woofers 2, 3, and 4 are tested individually with reversed phase (usually woofer 2 or 3 work best with phase reversed)
This simple approach produces surprisingly consistent results.
...
Cheers,
- James

Are there no problems when every subwoofer stand at different distances from the listener? How much difference can be tolerated?

[Kraniet]

Toole also argues against* placing bass around the room. One in every corner och one in the middle of every wall are some of the tips. Arguing that more bass speaker are better but above four is hitting diminish return.

Ingvar Öhman argues that the bass should preferably come from the front. That this would bring whats a more "pshycoacoustical correct wavefront".

Any comment in regard to this?

edit: sorry meant to say against not "for"

[Jacro]

Toole also argues against* placing bass around the room. One in every corner och one in the middle of every wall are some of the tips. Arguing that more bass speaker are better but above four is hitting diminish return.

Ingvar Öhman argues that the bass should preferably come from the front. That this would bring whats a more "pshycoacoustical correct wavefront".

Any comment in regard to this?

edit: sorry meant to say against not "for"

Hi Kraniet,

I believe that these two viewpoints, of Toole and Öhman, are both valid, each within a limited, optimized context.

The importance of maintaining a frontal derived wavefront is dependent on a number of parameters. One parameter set is the low pass corner frequency and/or low pass slope.

It works best if the low pass corner frequency is reduced and/or the low pass slope is increased, progressively for each woofer as the placement of the woofers is moved from front to beside the listener and then behind the listener. Also, the woofers are progressively reduced in level as they are place farther from the frontal launch region.

Each additional woofer is utilized mainly to provide a sequentially smaller and smaller correction to the modal reform. If this is done properly, there will be no audible difference in the perception of the appropriate psycho-acoustical wavefront direction.

All the parameters must be included before determining the ideal low pass frequencies, but in one example of an optimized room, the frontal woofers were crossed in at 122 Hz, the side woofers were crossed at 83 Hz and the rear woofers were crossed at 61 Hz. There were some additional optimized attributes but the system was carefully tested for tonal and spatial distortions relative to the addition of the side and rear woofers and was found to be transparent in this regard. Of course, the room parameters must support this type of low frequency architecture.

My more universal preference is to utilize a multiple woofer set that has all the individual woofers placed in the frontal hemisphere forward of the listening position, more inline with Mr. Öhman’s preference, but with a few additional enhancements.

My goal with the front hemisphere woofer arrangement, is to still address the most significant modal characteristics of the room, and this can be accomplished in the frontal portion of the room, but one must think three dimensionally.

This requires placing woofers in an arrangement that is not only horizontal, x-axis, but also to diversify in the y and z-axis, with at least one woofer above the half-height point of the room. With this approach, one can substantially accomplish the modal control of woofers placed on each of the four walls, while still maintaining 100% frontal wavelaunch.

In all of these approaches, one or more of the woofer units may contribute most effectively when operated with altered or reversed phase.

If one uses substantially full range (deep bass) left and right channel loudspeakers, then they can be included as one or two of the multi-sub elements.

I prefer to step down the gain of the left and right loudspeakers by 3 to 6 dB below the approximate frequency of the highest subwoofer crossover, and operate in a power-sharing manner below this frequency, both smoothing room modes AND increasing system maximum output capability.

Hopefully this provides at least a partial answer to your question.

Best regards,

- James

[paa]

Thanks James for your answer about directivety below 1kHz.
Now imagine that I manage to make loudspeakers that maintain directivety down to say 80Hz, and want to add some subwoofers to them:
If I place two subs in the front corners, they will be boundry coupled in their frequency range and have 90+90 degrees of dispersion which should be fine. But probably we play the subs in mono so now we have 180 degrees bass dispersion which is quite a bit more more than the theoretically perfect main speakers, how do we sort this out? And if we want one or two more subs to improve on room modes, how would all that effect frequency response vs power response in such a system?

paa,

My computer is in for repair for a few days, so I will have to wait to address this question more completely as soon as my computer is returned to me.

You ask a good question. Part of the answer is that it is most important to maintain the directivity down to the Schroeder frequency. Below the Schroeder frequency things change and to some degree, the sound source becomes the room modes themselves, as opposed to the loudspeaker, but that is a concept that needs some further explanation.

I hope to get back to you with more on this topic in a few days.

All the best,

- James

Hello James,
Do you have any more information about the Schroeder frequency, and its importance in small rooms?

[Jacro]

Hi James,
Once again, thanks for your patience with all our questions.
However much I await more about these things I have a follow up question here.

Are there no problems when every subwoofer stand at different distances from the listener? How much difference can be tolerated?

Hello James,
Do you have any more information about the Schroeder frequency, and its importance in small rooms?

Hi paa,

The nature of sound sources when operating in the modal frequency region seems to be an issue of great confusion. It is fundamentally different than how systems operate, and what parameters are important, above the Schroeder frequency.

In the modal range, it may be more appropriate to think of the dominant source of the sound reaching your ears as being that of the modal output of the room, not the loudspeaker.

In most cases, the woofer drives the room and the modal nature of the room is what delivers the sound to your ears.

Our hearing requires a couple of cycles before it can determine a sound's timbre and when operating below the Schröeder frequency, reflections become an integral part of the sound almost instantly because the wavelengths are comparable to, or greater than, the room's dimensions.

So what we hear is no longer dominated by the loudspeaker but instead is created by the room boundaries and the sense of a perceived transient response is dominated by the amplitude smoothness and harmonics of the bass fundamentals.

There are at least two arrangements that can substantially override this characteristic of the modal frequency range;

1) a woofer system that maintains high directivity at modal frequencies (a rarity, but we have built, tested, and characterized these types of systems to determine their effectiveness)

2) using a near field woofer close to the listening seat, such that the dominant arrival is the direct arrival from the woofer itself (this is a superior technique not commonly used in the industry as of yet. I believe this approach is ahead of its time and will become a standard in another 5 to 10 years. We have developed systems using this approach that consistently outperform all other approaches)

In terms of differential distances between different woofers and the listener, this is not an audible issue from a phase/transient standpoint. If there are phase/timing errors, it is the amplitude response change due to phase differentials that is audible, not the phase difference itself. In fact, when distributing multiple subwoofers around the room, one of the useful techniques to create maximum smoothness in the frequency response, is to alter the phase of at least one of the woofers to adjust the modal development and amplitude summation among the woofers as they drive the room modes.

When setting up distributed woofer systems I tend to use lower frequency low-pass filters for woofers placed behind and/or farther from the listener than the frontal woofers, attempting to maintain less than a quarter wavelength differential between all woofers when measuring their distances to the listener.

As usual, this is a very limited explanation relative to all of the acoustic and psycho-acoustics effects in play, but hopefully this at least gives you some useful information on the topic.

Cheers,

- James

[Nattlorden]

My goal with the front hemisphere woofer arrangement, is to still address the most significant modal characteristics of the room, and this can be accomplished in the frontal portion of the room, but one must think three dimensionally.

This requires placing woofers in an arrangement that is not only horizontal, x-axis, but also to diversify in the y and z-axis, with at least one woofer above the half-height point of the room. With this approach, one can substantially accomplish the modal control of woofers placed on each of the four walls, while still maintaining 100% frontal wavelaunch.

I'm stacking my woofers in the front coners - floor to ceiling. Does this conform, or would you also say I'd need some inbetween?

And what's your view about stereo vs. mono for (sub)woofers, say crossed at 80Hz?

And also something completely different - as I take it you've been around a lot over the years - have you crossed paths with Albert Von Schweikert and if so, have you any stories or such to tell about it?

[Jacro]

My goal with the front hemisphere woofer arrangement, is to still address the most significant modal characteristics of the room, and this can be accomplished in the frontal portion of the room, but one must think three dimensionally.

This requires placing woofers in an arrangement that is not only horizontal, x-axis, but also to diversify in the y and z-axis, with at least one woofer above the half-height point of the room. With this approach, one can substantially accomplish the modal control of woofers placed on each of the four walls, while still maintaining 100% frontal wavelaunch.

I'm stacking my woofers in the front coners - floor to ceiling. Does this conform, or would you also say I'd need some inbetween?

And what's your view about stereo vs. mono for (sub)woofers, say crossed at 80Hz?

And also something completely different - as I take it you've been around a lot over the years - have you crossed paths with Albert Von Schweikert and if so, have you any stories or such to tell about it?

Hello Night Lord,

Your arrangement of creating a line source of woofers in the front corners is an excellent approach that can work very well in many types of rooms. It is one of the best at providing both smooth response, due to a diversity line wave room interface, AND, great large signal capability due to corner re-enforcement.

It is unlikely, but if for some reason the shape, dimensions or rigidity of your room still results in non-uniform amplitude response, you may try adding one more woofer, placed midway on the front wall or midway on a sidewall, to further smooth the response, but in most rooms this would not be necessary with your configuration.

As an alternative, the Welti approach of placing 4 woofers on the floor, centered on each of the walls, can be transposed to the front wall, with a woofer centered on the floor, the sidewalls, and the ceiling, all pressed against the front wall.

In terms of maintaining stereo signals in the woofer systems operating below 80 Hz, it depends on room size, modal conditions, and program characteristics.

It is rare that recordings will contain audible separation below 80 Hz but one can generate “artificial” signals that may exhibit slight audibility under controlled conditions in very large rooms, but this does not translate to meaningful differences in the modal range in standard living rooms/studios because the modal nature of the room is to cause the reflected signals to act as a left/right channel-summing network.

My preferred system is to use full range main left and right channel loudspeakers (down to at least 50 Hz, preferably down to below 40 Hz) and have them maintain the separate, stereo, signals, while further incorporating a monophonic, multiple woofer system architecture as a shared bass system below 100 Hz.

Relative to Albert Von Schweikert, I only have a vague memory of him calling a number of years ago to ask me for advise about some patent issues and questions about planar magnetic/ribbon transducers. I have never met him, so I apologize for not having something more intriguing to convey.

All the best,

- James

[paa]

James, how do you manage to have all these different crossover frequencies and shared frequency ranges to sum at something close to phase coherence?

[Jacro]

James, how do you manage to have all these different crossover frequencies and shared frequency ranges to sum at something close to phase coherence?

paa,

I believe I already addressed this in my previous answers, but maybe I wasn’t clear enough.

To summarize, first arrival phase is relevant to the extent it has summation effects on amplitude (either amplitude interaction with the main L/R loudspeaker and/or amplitude summation of the multiple woofers and modal interactions), but (assuming standard room size and crossover frequency well below the Schröeder frequency) the direct audibility of the phase of the direct sound from the woofer is not significant.

As I suggested, our hearing requires a couple of cycles before it can determine a sound's timbre and when operating below the Schröeder frequency and reflections are nearly instantaneous in becoming an integral part of the sound because the wavelengths are comparable to, or greater than, the room's dimensions.

It is difficult to understand intuitively, but the source of audible low frequency arrivals is dominated by the room itself, not the loudspeaker.

A useful analogy may be that of a bass reflex loudspeaker operating at the tuning frequency; the woofer has a direct output to the listener, but the sound heard by the listener is dominated by vent output due to the Helmholtz resonance transferring energy from the rear-side of the woofer diaphragm coupling through the resonant mode of the enclosure. The phase of these two sources is not ideal from the standpoint of the first arrival from the front side of the woofer diaphragm, but the vent dominates in a manner that overrides the direct sound of the woofer, making the phase relationship insignificant.

Similarly, below the Schröeder frequency the woofer output drives the modes of the room and the modal output of the room is the dominant arrival perceived by the ear-brain.

Again, the phase plays a role in this case, but the significance of the phase is its affect on amplitude. If you hear any anomaly related to the phase relationships not being ideal, in the normal sense, that anomaly is an amplitude error caused by the phase relationship, not the audibility of the phase error itself.

The above is not a perfect analogy, but possibly it provides a useful example of another system that has a dominant source that has an ‘indirect’ relationship to the woofer output reaching the listener.


All of this is why I started my first answer with this statement:
“The nature of sound sources when operating in the modal frequency region seems to be an issue of great confusion. It is fundamentally different than how systems operate, and what parameters are important, above the Schroeder frequency.”

The rules we are used to applying above the Schröeder frequency are not necessarily valid below the Schröeder frequency. It takes a lot of effort to sort these things out, partially because the belief systems we have established from rules that we apply to certain aspects of audio and therefore assume that we can intuitively transfer those rules to all other aspects of the system.

Also, as I stated previously, if one ‘decouples’ the woofers from the modal conditions of the room, such as using high directivity or near-field woofers, then the rules that apply above the Schroeder frequency can still have validity, but these two approaches are not usually applied.

Any clearer?

- James

[MagnusÖstberg]

Hi James!

Such low crossover frequency as 40-50hz puts large demands on the main to be able to move substantial amount of air.

Don´t you see the advantage of having the larger woofers operate up to 80-100hz as that definetly makes things easier for the mains who can concentrate on upper bass and above? Why put that extra stress on them?

[Jacro]

Hi James!

Such low crossover frequency as 40-50hz puts large demands on the main to be able to move substantial amount of air.

Don´t you see the advantage of having the larger woofers operate up to 80-100hz as that definetly makes things easier for the mains who can concentrate on upper bass and above? Why put that extra stress on them?

Hi MagnusÖstberg,

I agree with you.

What I had stated it previously, but I forgot to mention again in the last post, is that I always "shelf down" my mains by approximately 6 to 12 dB to reduce the load on the main loudspeakers, but keep them active at that reduced level from 80 or 100 Hz down to their lower limit, to provide a more effective integration with the subwoofers and to create a greater number of diverse sources of bass in the room for further improvements in smoothness.

The greater the number of subwoofers that are added, the greater the amount of LF amplitude reduction can be applied to the mains, reducing their output requirements.

Make sense?

- James

[paa]

I womder if this concept is similar to the products called "basstöd" by ino?

[Nattlorden]

James, thank you very much your reply.

[MagnusÖstberg]

I womder if this concept is similar to the products called "basstöd" by ino?

Not exactly as "Basstöd" is limmited to one woofer per channel, passively connected to the mains.

[Jacro]

I wonder if this concept is similar to the products called "basstöd" by ino?

paa,

I am not familiar with "basstöd".

Can you explain the concept?

Thank you,

- James

[MagnusÖstberg]

Hi James!

Such low crossover frequency as 40-50hz puts large demands on the main to be able to move substantial amount of air.

Don´t you see the advantage of having the larger woofers operate up to 80-100hz as that definetly makes things easier for the mains who can concentrate on upper bass and above? Why put that extra stress on them?

Hi MagnusÖstberg,

I agree with you.

What I had stated it previously, but I forgot to mention again in the last post, is that I always "shelf down" my mains by approximately 6 to 12 dB to reduce the load on the main loudspeakers, but keep them active at that reduced level from 80 or 100 Hz down to their lower limit, to provide a more effective integration with the subwoofers and to create a greater number of diverse sources of bass in the room for further improvements in smoothness.

The greater the number of subwoofers that are added, the greater the amount of LF amplitude reduction can be applied to the mains, reducing their output requirements.

Make sense?

- James

Hi!

I must admit I don´t fully understand the meaning of extending the mains further down than 70-100hz, as they still will give some output down to around 50-70hz, even using a 5th order acoustic roll-off. This still means that the bassmodules can be placed fairly free from the mains because of the long wavelenght without compromissing.

I would say that the greater numbers of subwoofers used we will get greater LF amplitude reduction in the subwoofer range, thus increased headroom in the bass range. And ofcourse the acoustic advantage one can achive using multiple woofers spread out. But there will be no reduction in stress for the mains using my way, that is true.

However, I like the idea of letting more stress off the mains. This will however, as I see it, result in special demands om both filter and woofers. Let us play with the thought of gradually reduce the main output under 300hz with 6db/octav down to 75hz where they start rolling-off with 30dB/octav. This would mean that the woofers would have to be able to play up to +600hz - and placement would be more critical.

All in all, I think this route is too complex and will not give bennefit in comparison with the eefort put in. Better to make sure the main can handle all the output you need as long as they are cut crossed over acustically with 5th order around 70-90hz.

What is your thoughts around this?

Best regads
Magnus

[celef]

2) using a near field woofer close to the listening seat, such that the dominant arrival is the direct arrival from the woofer itself (this is a superior technique not commonly used in the industry as of yet. I believe this approach is ahead of its time and will become a standard in another 5 to 10 years. We have developed systems using this approach that consistently outperform all other approaches)

i'm surprised none haven't commented the above, i'm not sure i understand the meaning of it but it sounds to me like the subwoofer placement in early home cinema setups, like coffee table subwoofers, i'm sure this is not what you're saying?

[Nattlorden]

I did think that was what he was saying.... but envisioned element facing listeners, not down as in most of those coffee-table subwoofers.

( And I've begun thinking if this placement wouldn't work fine for a couple of subs for my home theater setup (note for James - not the system described above, the tv/home theater room is a much smaller one upstairs) )

[paa]

I wonder if this concept is similar to the products called "basstöd" by ino?

paa,

I am not familiar with "basstöd".

Can you explain the concept?

Thank you,

- James

Basstöd may be translated to: "Bass Support".
It's a subwoofer connected with a special crossover together with the main speaker, and it may also have a more powerful driver than the main speaker bass driver.
However, I do not know all the details, maybe someone who know more can fill in the gaps.

[paa]

2) using a near field woofer close to the listening seat, such that the dominant arrival is the direct arrival from the woofer itself (this is a superior technique not commonly used in the industry as of yet. I believe this approach is ahead of its time and will become a standard in another 5 to 10 years. We have developed systems using this approach that consistently outperform all other approaches)

i'm surprised none haven't commented the above, i'm not sure i understand the meaning of it but it sounds to me like the subwoofer placement in early home cinema setups, like coffee table subwoofers, i'm sure this is not what you're saying?

I'm wondering if a listener moving relative to the closely placed subwoofer would upset the balance of the percieved frequency response.

[Jacro]

Okay Gentlemen,

It appears that there are no more questions.

I just wanted to say that I enjoyed interacting with you.

If there is anything more that you wish to discuss or if you feel there is another area in the forum that you would like me to participate in, please let me know.

All the best,

- James

PS - Also, if I missed anyone's question, I apologize. If there is something remaining unanswered, please let me know and I'll do my best to provide an answer.

[Jacro]

2) using a near field woofer close to the listening seat, such that the dominant arrival is the direct arrival from the woofer itself (this is a superior technique not commonly used in the industry as of yet. I believe this approach is ahead of its time and will become a standard in another 5 to 10 years. We have developed systems using this approach that consistently outperform all other approaches)

i'm surprised none haven't commented the above, i'm not sure i understand the meaning of it but it sounds to me like the subwoofer placement in early home cinema setups, like coffee table subwoofers, i'm sure this is not what you're saying?

I'm wondering if a listener moving relative to the closely placed subwoofer would upset the balance of the percieved frequency response.

paa,

That is a very good observation.

This one of the aspects that must be properly coordinated with the specific system conditions. It is rather complex to generalize in short discussion, but if one mounts the woofer close enough to the listener that it is within the theoretical "near-field" condition, then inverse-square-law does not apply and the system is not particularly sensitive to movement of the listener's head.

If one places the woofer somewhat farther away, just outside of the theoretical "near-field", then the head movement will not approach a great enough change in distance to significantly alter level, even when inverse-square-law is in effect, due to the ratio of head movement to total distance being quite small.

[[As you may know, “near field” is used as a general term in audio, meaning listening close to a sound source, such that the direct sound to reflected sound ratio is substantially elevated to the point that the environmental effects are minimal. This general version of ‘near field’ should actually have a different name, such as ‘close field’ or ‘direct field’ listening.

“Near-field” is actually a specifically defined acoustical term that is based the precise relationship of the dimensions of the emission source and frequency/wavelength. In the true “near-field”, the sound field maintains greater directivity and more consistent output vs. distance, than it does in the “far-field”.]]

When properly set up, this change in head movement will have an insignificant effect, but the advantages relative to elimination of modal effects will be quite impressive. Many consider the sonic effect on their systems to be quite transformational while substantially obviating the need for multiple subwoofers.

Of course, this approach is less satisfying for listening while wandering around the room, but for specified seated listening for up to 3 or 4 people, it can be quite effective.

Cheers,

- James

[Jacro]

Hi James!

Such low crossover frequency as 40-50hz puts large demands on the main to be able to move substantial amount of air.

Don´t you see the advantage of having the larger woofers operate up to 80-100hz as that definetly makes things easier for the mains who can concentrate on upper bass and above? Why put that extra stress on them?

Hi MagnusÖstberg,

I agree with you.

What I had stated it previously, but I forgot to mention again in the last post, is that I always "shelf down" my mains by approximately 6 to 12 dB to reduce the load on the main loudspeakers, but keep them active at that reduced level from 80 or 100 Hz down to their lower limit, to provide a more effective integration with the subwoofers and to create a greater number of diverse sources of bass in the room for further improvements in smoothness.

The greater the number of subwoofers that are added, the greater the amount of LF amplitude reduction can be applied to the mains, reducing their output requirements.

Make sense?

- James

Hi!

I must admit I don´t fully understand the meaning of extending the mains further down than 70-100hz, as they still will give some output down to around 50-70hz, even using a 5th order acoustic roll-off. This still means that the bassmodules can be placed fairly free from the mains because of the long wavelenght without compromissing.

I would say that the greater numbers of subwoofers used we will get greater LF amplitude reduction in the subwoofer range, thus increased headroom in the bass range. And ofcourse the acoustic advantage one can achive using multiple woofers spread out. But there will be no reduction in stress for the mains using my way, that is true.

However, I like the idea of letting more stress off the mains. This will however, as I see it, result in special demands om both filter and woofers. Let us play with the thought of gradually reduce the main output under 300hz with 6db/octav down to 75hz where they start rolling-off with 30dB/octav. This would mean that the woofers would have to be able to play up to +600hz - and placement would be more critical.

All in all, I think this route is too complex and will not give bennefit in comparison with the eefort put in. Better to make sure the main can handle all the output you need as long as they are cut crossed over acustically with 5th order around 70-90hz.

What is your thoughts around this?

Best regads
Magnus

Magnus,

I agree, that IF your main left/right loudspeakers cannot handle low frequencies without distortion, then the approach I am suggesting will not work properly. In the case of weak main loudspeakers I would recommend a few other approaches.

But, if your main loudspeakers are capable of full system output with low distortion down to 40 or 50 Hz (with frequencies below 100 Hz shelved down by at least 6 dB) then the room modes can be smoothed by using only one or two subwoofers and the sound quality will be very good.

If the main loudspeakers are weak at low frequencies and must be high passed at higher frequencies then 4 subwoofers carefully placed will usually be necessary.

Each approach has tradeoffs, but I prefer to have my trade-off be one of "quantity", not "quality". I can always improve "quantity" of output by incorporating mains or subwoofers that have higher output capability, but improving the sound "quality" of the final result is more difficult and requires specialized efforts.

I believe the complexity is reasonable, particularly since the improvement in quality is significant.

I hope this makes sense.

Let me know if I am still not addressing your question in the manner you are looking for.

- James

[paa]

James, would you think the HOM reduction foam also works as an acoustic lens that further smothes the dispersion whithin the desired coverage angle?

Interesting idea.

The current goal of applying foam to a waveguide to reduce HOMs is to be able to maintain consistent absorption such that directivity is not changed from the ideal directivity of the waveguide. Correctly applied, directivity does not change when the foam is applied.

Attempting to use the foam to change directivity does add a degree of complexity that may be difficult to fully characterize. As one changes absorption rates, it must be achieved in a manner that the absorption discontinuities don't increase HOMs, simulating diffractive effects.

That said, the concept of using the foam to alter directivity while reducing HOMs may be a very good idea. The ability to provide an addition variable to control directivity in waveguides may allow one to relax the directivity capability of a waveguide in order to achieve an improvement of some other parameter (such as efficiency, or bandwidth, etc.), and then re-optimize the directivity with a variable density foam insert.

Definitely worth exploring.

- James

I believe that acoustical transparent foam (now often called "filter foam") was claimed to behave like an acoustical lens, expanding the directivety of narrow beaming tweeters, when it was introduced in the seventies. To what extent that really worked I have no idea.

[Jacro]

Okay Gentlemen,

It appears that there are no more questions.

I just wanted to say that I enjoyed interacting with you.

If there is anything more that you wish to discuss or if you feel there is another area in the forum that you would like me to participate in, please let me know.

All the best,

- James

PS - Also, if I missed anyone's question, I apologize. If there is something remaining unanswered, please let me know and I'll do my best to provide an answer.

[jeppe]

Hello James, I hope you still visit faktiskt every now and then.

I´ve been living with (the clue) now for about a month and I just want to say a few words about them. They are not good or very good but utterly brilliant!

I bought them unheard and now I can´t stop listening to them.
thanks!!

[celef]

jeppe, berätta gärna hur du gjorde för att komma över ett par! och vad fick dig att köpa olyssnat? beställde fredrik ett par oxå?

[jeppe]

mailade Sjöfn och fick det glädjande beskedet att dom har sänkts till 999 dollars och exporteras direkt från Sjöfn hifi, alltså inga omvägar via butik

Tog väl en vecka tills jag hade dom. Kostade allt som allt (ink moms) 7700 sek. Vet inte om Fredrik har beställt.

Varför köpa olyssnat? tja, nyfikenhet.. jag var ganska säker på att dom skulle vara precis så pass bra som jag tycker att dom är

[celef]

tack för svar! nyfiken som jag är: köpte du högtalarna på grund av skriverierna här eller skriverier någon annanstans, typ i hifimagasin eller andra internetforum?

[jeppe]

hmm, ja olika forum och recensioner stärkte köpbeslutet sen var det mycket hur konstruktören kan förklara varje aspekt av konstruktionen och tänket bakom lösningarna.

[Nattlorden]

hmm, ja olika forum och recensioner stärkte köpbeslutet sen var det mycket hur konstruktören kan förklara varje aspekt av konstruktionen och tänket bakom lösningarna.

[Jacro]

Hello James, I hope you still visit faktiskt every now and then.

I´ve been living with (the clue) now for about a month and I just want to say a few words about them. They are not good or very good but utterly brilliant!

I bought them unheard and now I can´t stop listening to them.
thanks!!

Hello Jeppe,

Thank you for your enthusiastic comments!

(the clue) has been a rewarding project. It is one of the lowest priced loudspeakers I have ever designed, but by incorporating the novel sound field construct, it provides advantages over more expensive approaches.

Most of all, we wanted (the clue) owners to have more fun with music, so I'm very happy to hear that you are enjoying your new loudspeakers.

Cheers,

- James

[metheny]

(the clue) has been a rewarding project. It is one of the lowest priced loudspeakers I have ever designed, but by incorporating the novel sound field construct, it provides advantages over more expensive approaches.

Most of all, we wanted (the clue) owners to have more fun with music, so I'm very happy to hear that you are enjoying your new loudspeakers.

Cheers,

- James

Hi, sorry for not having read the whole thread, the answer might already have been given. But I'm wondering if the elements are of custom design or "off-the-shelf"? Will it be possible to buy the speaker as a kit, i.e just the filter components and the elements together with instructions on how to build the boxes?

[_Fredrik_]

hmm, ja olika forum och recensioner stärkte köpbeslutet sen var det mycket hur konstruktören kan förklara varje aspekt av konstruktionen och tänket bakom lösningarna.

Fy fan va fett, grattis. Nej jag har inte beställt, jag har väntat på att du ska göra det först.

Va roligt att du är nöjd!

[Jacro]

(the clue) has been a rewarding project. It is one of the lowest priced loudspeakers I have ever designed, but by incorporating the novel sound field construct, it provides advantages over more expensive approaches.

Most of all, we wanted (the clue) owners to have more fun with music, so I'm very happy to hear that you are enjoying your new loudspeakers.

Cheers,

- James

Hi, sorry for not having read the whole thread, the answer might already have been given. But I'm wondering if the elements are of custom design or "off-the-shelf"? Will it be possible to buy the speaker as a kit, i.e just the filter components and the elements together with instructions on how to build the boxes?

Hi Metheny,

The tweeter/waveguide and woofer elements are proprietary (not off the shelf). A kit is not available at this time.

To replicate the performance of a production version of (the clue) is difficult to realize in a kit, due to the final "in-box" calibration process. Currently, it would cost more to produce a high-performance kit than a standard production unit, as we would have to build the kit into an enclosure, complete the final calibration process and then disassemble the elements and package them in kit form.

(the clue) wasn't designed with the idea of making a kit version of the same performance ability.

That said, if there are enough requests, we will revisit the exploration of how to make a kit with the same sound quality as the standard version.

Thank you for your interest. I will add +1 to our kit request list.

All the best,

- James

[luminous]

Hi Jacro,

I have been reading your posts in this thread with great interest. I was wondering if you would like to share your view on this topic:

How can you tell, by taking measurements, that you have achieved a neutral perceived timbre (flat perceived frequency response) in a room?

Most people seems to agree that the goal isn't just a flat in-room frequency response, but there doesn't seem to be a generally accepted answer to this question either.

Regards, Viktor

[jeppe]

hmm, ja olika forum och recensioner stärkte köpbeslutet sen var det mycket hur konstruktören kan förklara varje aspekt av konstruktionen och tänket bakom lösningarna.

Fy fan va fett, grattis. Nej jag har inte beställt, jag har väntat på att du ska göra det först.

Va roligt att du är nöjd!

Tack! jodå väldigt nöjd, dom är verkligen plug and play i mitt rum. Har aldrig hört mitt usla rum ljuda så fint. Har du beställt nu då?!

[Ragnwald]

jeppe, vad fick du betala allt som allt, för ett par?

[Audix]

Jeppe, var i sverige finns du? Vill du ta emot nyfikna besökare?

[Lust]

jeppe, vad fick du betala allt som allt, för ett par?

mailade Sjöfn och fick det glädjande beskedet att dom har sänkts till 999 dollars och exporteras direkt från Sjöfn hifi, alltså inga omvägar via butik.

Tog väl en vecka tills jag hade dom. Kostade allt som allt (ink moms) 7700 sek.

[Ragnwald]

Ok, tack.

[jeppe]

Jeppe, var i sverige finns du? Vill du ta emot nyfikna besökare?

Bor i närheten av Tumba i Stockholm men jag är mitt i en separation så det är tyvärr inte aktuellt med besök

[metheny]

Har någon jämfört dessa med piP?

[metheny]

mailade Sjöfn och fick det glädjande beskedet att dom har sänkts till 999 dollars och exporteras direkt från Sjöfn hifi, alltså inga omvägar via butik [..] Kostade allt som allt (ink moms) 7700 sek.

$999 ~ 6300 SEK, moms 25 %, ~1500 kr, och då är inte frakten medräknad. Hur har du räknat? Skickas de utan fraktkostnad?

[Lust]

Bor i närheten av Tumba i Stockholm men jag är mitt i en separation så det är tyvärr inte aktuellt med besök

Det är då faen vad svårt många kvinnor har med nya högtalare ...

[meanmachine]

Hi Jarco..

After have read this thread and with special interest your posts I admire you're humble but convincing way of exressing your self.

I'ts above all fantastic to have you here and that you even bother to takte the time with us is just marvellous. That tells me one thing paticular and that is that You really love this hobby of ours. What else can bring a man of your reusume and dignity to chat on a forum, and humbly but ferm guide lost souls with revelaing thuths...

Really impressive of you. It shows You're a great man with a great mind. Just by that I'm very interested in The clue.

Tnx for the time you've spent here and all the best.


Hope you forgive if I spell wrong.

[Jacro]

Hi Jacro,

I have been reading your posts in this thread with great interest. I was wondering if you would like to share your view on this topic:

How can you tell, by taking measurements, that you have achieved a neutral perceived timbre (flat perceived frequency response) in a room?

Most people seems to agree that the goal isn't just a flat in-room frequency response, but there doesn't seem to be a generally accepted answer to this question either.

Regards, Viktor

Hi Viktor,

Your question points to what may be the most sonically significant set of loudspeaker parameters, which are those that describe the power response of a sonic emission device.

A rule of thumb guideline is probably what is wanted, but unfortunately, in this case, an over simplification would not lead to an effective method.

That said, I'll provide some of the basics in the approach that works and hopefully, the ideas will be of some value. But, I have found that it is not possible to have a fully useful transmission of knowledge in this medium.


To achieve a proper result takes a much more rigorous effort than most folks are interested in pursuing. It is a tedious process, but the results are well worth the effort.

While it is clear what is required, it is a multi-dimensional issue that is quite complex to accomplish.

Terms like "in-room-response" are really far too vague to really explain the actual complexity of the multi-dimensional issue that needs to be resolved.

To disclose how to achieve the proper result is difficult to accurately convey in forum discussion, partly because it is a dynamic process that varies depending on interacting contexts, which would require support of a pictorial presentation of the multi-feedback and correction approach.

Ultimately, one must arrive at a power response emitted from the loudspeaker system (the amplitude/response curve “emitted IN each direction”) that creates the “head related power response” (amplitude at each frequency “received FROM” each direction) that is appropriate to create timbral and spatial neutrality.

There are well-defined vector/amplitude sets that provide the most neutral timbre, but they are inextricably related to the use configuration model of the loudspeaker device.

This includes the boundary relationship of the transducer, including 'global' listening environment boundaries and 'local' enclosure boundaries.

If the device is boundary coupled to the front wall, floor, sidewall, or free standing, requires a very different radiation pattern. And each of these placement models has a range of parameters to be defined, again related to the environment and enclosure relationships (front of enclosure to boundary, vs. back and sides of enclosure to boundaries).

The use model also includes the relationship of the listener’s upper body to the transducer/enclosure, such as angle (vertical and horizontal), distance, etc.

Example: Depending on wave launch source position relationship to listener, the optimal power response will differ.

This is just a small subset of the variations that must be taken in to consideration.

Unfortunately, most loudspeaker manufacturers still don’t define a specific use model (because they want to let people use the speakers where ever it is convenient) so for most loudspeakers, it is not possible to accurately define an effective power response.

Each use model, once clearly defined, has a unique optimal radiation pattern/power response.

Once the use model and boundary relationship and listener relationship is defined then a specialized set of measurements must be taken to calibrate the power response.

To be truly effective, this is not defined by just measuring the radiation pattern outward from the loudspeaker.

One approach is to measure the polar response outward from the device under test, with a more precision angle weighting function depending on angle, with certain angles tangent to the listening position differentiated from all other angles with a specific non-linear priority.

Then, with a varying boundary set (multiple room sizes and forms) one measures a “reception” power response at the listener’s torso (upper body).

Ultimately, while measuring the power response “IN all directions” from the loudspeaker, one is really only providing a beginning step in defining what is more important, which is what is the amplitude response of arrivals at the listener “FROM all directions”.

By having a defined set of amplitude vs. frequency arrivals “from each direction” one can work back to the loudspeaker device being calibrated for a power response of how it should radiate “in all directions”.

This is combined with a few other adjustments and design processes that impact the sound field around the listener’s head.

To make the matter even more complex, the ideal power response is not a single channel, emission definition, but must be calibrated to the channel count employed to achieve an optimal spatial response while maintaining timbral neutrality. This requires the power response from each loudspeaker to be recalibrated differently for “stereo” if timbre is to be maintained and “multi-channel coloration” minimized.

Then, the next step, depending on the architecture of the loudspeaker, ideally, one optimizes the polar response(s) to maintain neutrality across a listening window wide enough for at least three listeners seated beside each other. Some loudspeaker system topologies allow for this adaptation more than others.

Again, depending on the type of emission architecture (dipole, monopole, free-standing, ½ space, ¼ space..., etc.) and use model (listener/loudspeaker/environment relationship) the above stated calibration technique will result in a unique amplitude/vector set for each system type.

Hopefully, what I have written so far is of interest, even though it does not provide a quick and simple answer to the very important question that was asked.

I’ll see if there is feedback on what I’ve written so far, and if there is interest, we can explore further towards a more complete answer.

Cheers,

- James

[celef]

i really like your posts jacro and i'm glad you still visiting us!

whats your opinion of all those loudspeaker measruements that stereophile and soundstage are providing, are they a helpful tool for sorting out poorly deisgned speakers or are they just confusing for most readers?

[Ragnwald]

The thread has rendered to be very long, so i hope my question not have been up before.
As we see in this picture, there is an ideal angle against the wall, but i want to know ideal distance between speakers, and the ideal listening position? And have distance to sidewalls any influence?

[luminous]

Hi James,

Since your answer is quite long I have made quite many quotes below:

Ultimately, one must arrive at a power response emitted from the loudspeaker system (the amplitude/response curve “emitted IN each direction”) that creates the “head related power response” (amplitude at each frequency “received FROM” each direction) that is appropriate to create timbral and spatial neutrality.

How do you define spatial neutrality? The spatial aspects of reproduction (sensation of room, envelopement, liveliness etc) ought to be rather subject to individual taste, and most recordings are not meant for "sound field reproduction" but to create a pleasurable experience...

There are well-defined vector/amplitude sets that provide the most neutral timbre, but they are inextricably related to the use configuration model of the loudspeaker device.

This includes the boundary relationship of the transducer, including 'global' listening environment boundaries and 'local' enclosure boundaries.

If the device is boundary coupled to the front wall, floor, sidewall, or free standing, requires a very different radiation pattern. And each of these placement models has a range of parameters to be defined, again related to the environment and enclosure relationships (front of enclosure to boundary, vs. back and sides of enclosure to boundaries).

Do you mean that the "reference sound field" at the listening position differs with the use model?
Of course, the loudspeaker design would differ according to the loudspeaker use model if the same reference sound field at the listening position is the goal.

And by "There are well-defined vector/amplitude sets that provide the most neutral timbre", I don't know if I interpret vector/amplitude sets correctly but to define a sound field at the listener that provides the most neutral timbre, I guess one would need to have knowledge of how to go from "sound amplitude versus frequency versus angle of incidence versus time of incidence at the head" to perceived timbre, sounds like a very difficult problem to me.

The use model also includes the relationship of the listener’s upper body to the transducer/enclosure, such as angle (vertical and horizontal), distance, etc.

Example: Depending on wave launch source position relationship to listener, the optimal power response will differ.

I guess it depends on what the goal of the source is, but if it is to generate the same perceived timbre regardless of source angle to the listener, then why should the design goal be different for different source angles? After all our brain should be expecting all modifications on the sound field that our body gives for different sound angles.

This is just a small subset of the variations that must be taken in to consideration.

Unfortunately, most loudspeaker manufacturers still don’t define a specific use model (because they want to let people use the speakers where ever it is convenient) so for most loudspeakers, it is not possible to accurately define an effective power response.

I agree that it is a pity that not more manufacturers define a "use model" for their loudspeakers. They will never have full control over how the speakers will sound in peoples homes.

Each use model, once clearly defined, has a unique optimal radiation pattern/power response.

Once the use model and boundary relationship and listener relationship is defined then a specialized set of measurements must be taken to calibrate the power response.

To be truly effective, this is not defined by just measuring the radiation pattern outward from the loudspeaker.

One approach is to measure the polar response outward from the device under test, with a more precision angle weighting function depending on angle, with certain angles tangent to the listening position differentiated from all other angles with a specific non-linear priority.

Then, with a varying boundary set (multiple room sizes and forms) one measures a “reception” power response at the listener’s torso (upper body).

Do you take actual measurements at a listening position with a listener present? How do you know what such measurements should look like?

Ultimately, while measuring the power response “IN all directions” from the loudspeaker, one is really only providing a beginning step in defining what is more important, which is what is the amplitude response of arrivals at the listener “FROM all directions”.

By having a defined set of amplitude vs. frequency arrivals “from each direction” one can work back to the loudspeaker device being calibrated for a power response of how it should radiate “in all directions”.

This is combined with a few other adjustments and design processes that impact the sound field around the listener’s head.

I think the most interesting problem to discuss here is what kind of sound field you want to have at the listening location - if you know that, then it's a separate problem how you should design loudspeakers for a certain use model to achieve this.

To make the matter even more complex, the ideal power response is not a single channel, emission definition, but must be calibrated to the channel count employed to achieve an optimal spatial response while maintaining timbral neutrality. This requires the power response from each loudspeaker to be recalibrated differently for “stereo” if timbre is to be maintained and “multi-channel coloration” minimized.

This is an interesting subject as well… Multiple loudspeakers will of course add in a complex manner. I imagine that they will add differently at a microphone than at a listener's head. And it should also depend on the music signal - if the sound is correlated or uncorrelated between the channels.

Then, the next step, depending on the architecture of the loudspeaker, ideally, one optimizes the polar response(s) to maintain neutrality across a listening window wide enough for at least three listeners seated beside each other. Some loudspeaker system topologies allow for this adaptation more than others.

Again, depending on the type of emission architecture (dipole, monopole, free-standing, ½ space, ¼ space..., etc.) and use model (listener/loudspeaker/environment relationship) the above stated calibration technique will result in a unique amplitude/vector set for each system type.

Hopefully, what I have written so far is of interest, even though it does not provide a quick and simple answer to the very important question that was asked.

I’ll see if there is feedback on what I’ve written so far, and if there is interest, we can explore further towards a more complete answer.

For my part there is a large interest in these questions. Thanks for the meaty post!

/Viktor

[Jacro]

i really like your posts jacro and i'm glad you still visiting us!

whats your opinion of all those loudspeaker measruements that stereophile and soundstage are providing, are they a helpful tool for sorting out poorly deisgned speakers or are they just confusing for most readers?

Hi celef,

Thank you for your kind comments.

I don’t like to be critical of other’s approaches, but for me, the Stereophile "spatial averaging" looses too much specificity about the angular responses and has very little usefulness for anticipating the sound quality of the device under test.

The SoundStage approach provides more useful information, at least for a first approximation analysis, but, without knowing how to interpret the information, it may still be confusing for many readers.

Cheers,

- James

[Jacro]

The thread has rendered to be very long, so i hope my question not have been up before.
As we see in this picture, there is an ideal angle against the wall, but i want to know ideal distance between speakers, and the ideal listening position? And have distance to sidewalls any influence?

Hi Ragnwald,

Here are some of the general instructions from (the clue) User Guide. I included mainly the elements you asked about.

These are minimalist instructions to get the fundamental capability from (the clue). I can provide additional instruction to refine the performance further, if there is interest.

Let me know if you have further questions, or if you (or anyone else) would like any of (the clue) information sent directly to an email address.

All the best,

- James

1. Locate the loudspeakers on stands that are between approximately 19 and 22 inches in height.

2. Locate the loudspeakers within less than 2.5 inches of the wall behind them: the closer the better – but not actually touching the wall.

3. Position the loudspeakers so that the ratio of the distance between them (measured center-to-center) to the distance to your ears at the listening position is approximately 1 to 1.18.
(Example: 10.0 feet apart, center-to-center, AND 11.8 feet from a point halfway between the fronts of the two speakers, to your listening position.)

While the 1:1.18 ratio works well in most listening spaces, you might experiment with moving your listening spot a tad closer or further away; your room acoustics might be such that a different ratio affords even better sound.

4. The loudspeakers are designed to be toed-in at an angle of about 22.5 degrees. If they’re angled correctly, when you’re sitting in your official listening position, you’ll just barely be able to see the outer faces of the speakers.

5. At least one of the loudspeakers should have the center of its woofer more than 28 inches from a side wall.

7. Remove the grille boards: Like most loudspeakers, (the clue) performs better without them. If you have active pets, small children, or the occasional wobbly Saturday night, you can easily replace the grilles while you’re not listening (or just have on background music).

8. Acoustic damping material, approximately 2 to 2.5 inches thick, placed against the wall behind the speaker, will bring you even a bit more into the vaunted “you are there” listening experience.

Ideally, the damping material will extend at least 4 inches above the top of the loudspeaker, and from the back corner of the enclosure nearer the wall behind the loudspeaker inward towards the opposite loudspeaker by 24 inches or more, and down to within at least the bottom edge of the cabinet. This would be a piece that is 2” D x 18” H x24” W. Extending closer to the floor and widening by another 6” to 12” will afford slight improvement - but at diminishing returns.

9. Best sonic balance is achieved at a listening distance of between four and fourteen feet.

10. The loudspeakers are built acoustically to power rooms of up to around 2,200 cubic feet. If your listening room is larger than that, please contact Sjöfn Hi-Fi so that we can clue you in on solutions we’ve developed for positioning and enhancement in more spacious digs.

[Jacro]

Hi James,

Since your answer is quite long I have made quite many quotes below:

Ultimately, one must arrive at a power response emitted from the loudspeaker system (the amplitude/response curve “emitted IN each direction”) that creates the “head related power response” (amplitude at each frequency “received FROM” each direction) that is appropriate to create timbral and spatial neutrality.

How do you define spatial neutrality? The spatial aspects of reproduction (sensation of room, envelopement, liveliness etc) ought to be rather subject to individual taste, and most recordings are not meant for "sound field reproduction" but to create a pleasurable experience...

There are well-defined vector/amplitude sets that provide the most neutral timbre, but they are inextricably related to the use configuration model of the loudspeaker device.

This includes the boundary relationship of the transducer, including 'global' listening environment boundaries and 'local' enclosure boundaries.

If the device is boundary coupled to the front wall, floor, sidewall, or free standing, requires a very different radiation pattern. And each of these placement models has a range of parameters to be defined, again related to the environment and enclosure relationships (front of enclosure to boundary, vs. back and sides of enclosure to boundaries).

Do you mean that the "reference sound field" at the listening position differs with the use model?
Of course, the loudspeaker design would differ according to the loudspeaker use model if the same reference sound field at the listening position is the goal.

And by "There are well-defined vector/amplitude sets that provide the most neutral timbre", I don't know if I interpret vector/amplitude sets correctly but to define a sound field at the listener that provides the most neutral timbre, I guess one would need to have knowledge of how to go from "sound amplitude versus frequency versus angle of incidence versus time of incidence at the head" to perceived timbre, sounds like a very difficult problem to me.

The use model also includes the relationship of the listener’s upper body to the transducer/enclosure, such as angle (vertical and horizontal), distance, etc.

Example: Depending on wave launch source position relationship to listener, the optimal power response will differ.

I guess it depends on what the goal of the source is, but if it is to generate the same perceived timbre regardless of source angle to the listener, then why should the design goal be different for different source angles? After all our brain should be expecting all modifications on the sound field that our body gives for different sound angles.

This is just a small subset of the variations that must be taken in to consideration.

Unfortunately, most loudspeaker manufacturers still don’t define a specific use model (because they want to let people use the speakers where ever it is convenient) so for most loudspeakers, it is not possible to accurately define an effective power response.

I agree that it is a pity that not more manufacturers define a "use model" for their loudspeakers. They will never have full control over how the speakers will sound in peoples homes.

Each use model, once clearly defined, has a unique optimal radiation pattern/power response.

Once the use model and boundary relationship and listener relationship is defined then a specialized set of measurements must be taken to calibrate the power response.

To be truly effective, this is not defined by just measuring the radiation pattern outward from the loudspeaker.

One approach is to measure the polar response outward from the device under test, with a more precision angle weighting function depending on angle, with certain angles tangent to the listening position differentiated from all other angles with a specific non-linear priority.

Then, with a varying boundary set (multiple room sizes and forms) one measures a “reception” power response at the listener’s torso (upper body).

Do you take actual measurements at a listening position with a listener present? How do you know what such measurements should look like?

Ultimately, while measuring the power response “IN all directions” from the loudspeaker, one is really only providing a beginning step in defining what is more important, which is what is the amplitude response of arrivals at the listener “FROM all directions”.

By having a defined set of amplitude vs. frequency arrivals “from each direction” one can work back to the loudspeaker device being calibrated for a power response of how it should radiate “in all directions”.

This is combined with a few other adjustments and design processes that impact the sound field around the listener’s head.

I think the most interesting problem to discuss here is what kind of sound field you want to have at the listening location - if you know that, then it's a separate problem how you should design loudspeakers for a certain use model to achieve this.

To make the matter even more complex, the ideal power response is not a single channel, emission definition, but must be calibrated to the channel count employed to achieve an optimal spatial response while maintaining timbral neutrality. This requires the power response from each loudspeaker to be recalibrated differently for “stereo” if timbre is to be maintained and “multi-channel coloration” minimized.

This is an interesting subject as well… Multiple loudspeakers will of course add in a complex manner. I imagine that they will add differently at a microphone than at a listener's head. And it should also depend on the music signal - if the sound is correlated or uncorrelated between the channels.

Then, the next step, depending on the architecture of the loudspeaker, ideally, one optimizes the polar response(s) to maintain neutrality across a listening window wide enough for at least three listeners seated beside each other. Some loudspeaker system topologies allow for this adaptation more than others.

Again, depending on the type of emission architecture (dipole, monopole, free-standing, ½ space, ¼ space..., etc.) and use model (listener/loudspeaker/environment relationship) the above stated calibration technique will result in a unique amplitude/vector set for each system type.

Hopefully, what I have written so far is of interest, even though it does not provide a quick and simple answer to the very important question that was asked.

I’ll see if there is feedback on what I’ve written so far, and if there is interest, we can explore further towards a more complete answer.

For my part there is a large interest in these questions. Thanks for the meaty post!

/Viktor

Viktor,

Great questions!

These issues require more than quick answers. I am completing a couple projects with hard deadlines, so I will return to answer your questions as soon as I have some time to devote to them. I'll probably need to address one or two at a time.

All the best,

- James

[Nattlorden]

Clever. If I'd make speakers, and angling the front like NHT did isn't a good idea for some technical reasons, I'd consider making the box angled at the back... so placing them flat against the wall would put them at the correct angle.

Reading the placement instruction, I wonder if James and Ingvar shouldn't join forces, they seem to be on the (reasonably) same track.

[Jacro]

Clever. If I'd make speakers, and angling the front like NHT did isn't a good idea for some technical reasons, I'd consider making the box angled at the back... so placing them flat against the wall would put them at the correct angle.

Reading the placement instruction, I wonder if James and Ingvar shouldn't join forces, they seem to be on the (reasonably) same track.

Hi Nattlorden,

There may be some confusion on this issue. While the angle to the front wall as shown is a good starting point, the angle to the listener's ears is actually the final calibration step which, in some cases, alters the on-wall angle.

Original prototypes of (the clue) had angled surfaces to facilitate coupling the enclosure to the wall, as you suggest, but there were variables in some installations that still required angle adjustments, so the final enclosure was redesigned as a more standard configuration.

If one does consider angling the enclosure, it would be better to angle the back panel, as you suggest, instead of the front panel, as per NHT, unless one addresses the hyper-diffraction and baffle support collapse issues of the sharp, greater than 90-degree outer edge baffle angles.

Cheers,

- James

[Glebster]

1. Locate the loudspeakers on stands that are between approximately 19 and 22 inches in height.

2. Locate the loudspeakers within less than 2.5 inches of the wall behind them: the closer the better – but not actually touching the wall.

3. Position the loudspeakers so that the ratio of the distance between them (measured center-to-center) to the distance to your ears at the listening position is approximately 1 to 1.18.
(Example: 10.0 feet apart, center-to-center, AND 11.8 feet from a point halfway between the fronts of the two speakers, to your listening position.)

While the 1:1.18 ratio works well in most listening spaces, you might experiment with moving your listening spot a tad closer or further away; your room acoustics might be such that a different ratio affords even better sound.

4. The loudspeakers are designed to be toed-in at an angle of about 22.5 degrees. If they’re angled correctly, when you’re sitting in your official listening position, you’ll just barely be able to see the outer faces of the speakers.

5. At least one of the loudspeakers should have the center of its woofer more than 28 inches from a side wall.

7. Remove the grille boards: Like most loudspeakers, (the clue) performs better without them. If you have active pets, small children, or the occasional wobbly Saturday night, you can easily replace the grilles while you’re not listening (or just have on background music).

8. Acoustic damping material, approximately 2 to 2.5 inches thick, placed against the wall behind the speaker, will bring you even a bit more into the vaunted “you are there” listening experience.

Ideally, the damping material will extend at least 4 inches above the top of the loudspeaker, and from the back corner of the enclosure nearer the wall behind the loudspeaker inward towards the opposite loudspeaker by 24 inches or more, and down to within at least the bottom edge of the cabinet. This would be a piece that is 2” D x 18” H x24” W. Extending closer to the floor and widening by another 6” to 12” will afford slight improvement - but at diminishing returns.

9. Best sonic balance is achieved at a listening distance of between four and fourteen feet.

10. The loudspeakers are built acoustically to power rooms of up to around 2,200 cubic feet. If your listening room is larger than that, please contact Sjöfn Hi-Fi so that we can clue you in on solutions we’ve developed for positioning and enhancement in more spacious digs.

Geee, it sounds like you are dictating the 'user guide' of Ino/Guro speakers. Rather interesting how two persons sharing the same interest and, at least to some extent, sound philosophy can come up with the same user guidelines like 8000km apart, then again, maybe not!

[jeppe]

The thread has rendered to be very long, so i hope my question not have been up before.
As we see in this picture, there is an ideal angle against the wall, but i want to know ideal distance between speakers, and the ideal listening position? And have distance to sidewalls any influence?

I can provide additional instruction to refine the performance further, if there is interest.

Let me know if you have further questions, or if you (or anyone else) would like any of (the clue) information sent directly to an email address.

All the best,

- James

Yes, please send me further instructions: jrudefors@hotmail.com
appreciate it, thanks!

[luminous]

Viktor,

Great questions!

These issues require more than quick answers. I am completing a couple projects with hard deadlines, so I will return to answer your questions as soon as I have some time to devote to them. I'll probably need to address one or two at a time.

All the best,

- James

James,

No problem, I'll be waiting though. Choose the questions that you find the most interesting.

Best / Viktor

[Ragnwald]

9. Best sonic balance is achieved at a listening distance of between four and fourteen feet.

Thanks for a complete answer.
Is there no problem listening to a waveguide speaker at a distance of only 4 feet (1 meter)?
What is your opinion about the absolute ideal distance?

[Jacro]

Hi Jarco..

After have read this thread and with special interest your posts I admire you're humble but convincing way of exressing your self.

I'ts above all fantastic to have you here and that you even bother to takte the time with us is just marvellous. That tells me one thing paticular and that is that You really love this hobby of ours. What else can bring a man of your reusume and dignity to chat on a forum, and humbly but ferm guide lost souls with revelaing thuths...

Really impressive of you. It shows You're a great man with a great mind. Just by that I'm very interested in The clue.

Tnx for the time you've spent here and all the best.


Hope you forgive if I spell wrong.

Hi MeanMachine,

Thank you for your generous words.

After many years in audio, I still enjoy waking up every morning excited to continue working on the problem of recreating the experience of live music.

And it is great to interact with such an enthusiastic and informed group of audiophiles as I find here on Faktiskt.

For reasons I still don't understand, since my earliest days in audio I have always found the audio explorations of northern Europe more interesting than the USA or other parts of the world.

So, even though I don't have time for reading and writing in forums as much as I would like, I always enjoy taking the time to come back here.

Thanks again for the warm welcome,

- James

[_Fredrik_]

Dear Jacro, Faktiskt.se is the best hifi forum in the world! Yes it's true We have people like Nattis, eNils, celef, paa and Falkis. Great guys IMHO.

[roren]

Hi Jacro,

First I want to say that it's nice to have you here.
I find your ideas interesting and the the way that you use the drivers in the 'clue' is
new to me. But there's allways some tradeoffs. So now I just want to ask, What do
you feel is the biggest sacrifice you have made when having a wideband driver?
How capabale is the clue when it comes to soundpressure?

Sometimes I just like to play it loud.

Rolf

[celef]

Yes it's true We have people like Nattis, eNils, celef, paa and Falkis. Great guys IMHO.

you are so sweet, i dance for you

[Jacro]

Hi James,

Since your answer is quite long I have made quite many quotes below:

Ultimately, one must arrive at a power response emitted from the loudspeaker system (the amplitude/response curve “emitted IN each direction”) that creates the “head related power response” (amplitude at each frequency “received FROM” each direction) that is appropriate to create timbral and spatial neutrality.

How do you define spatial neutrality? The spatial aspects of reproduction (sensation of room, envelopement, liveliness etc) ought to be rather subject to individual taste, and most recordings are not meant for "sound field reproduction" but to create a pleasurable experience...

There are well-defined vector/amplitude sets that provide the most neutral timbre, but they are inextricably related to the use configuration model of the loudspeaker device.

This includes the boundary relationship of the transducer, including 'global' listening environment boundaries and 'local' enclosure boundaries.

If the device is boundary coupled to the front wall, floor, sidewall, or free standing, requires a very different radiation pattern. And each of these placement models has a range of parameters to be defined, again related to the environment and enclosure relationships (front of enclosure to boundary, vs. back and sides of enclosure to boundaries).

Do you mean that the "reference sound field" at the listening position differs with the use model?
Of course, the loudspeaker design would differ according to the loudspeaker use model if the same reference sound field at the listening position is the goal.

And by "There are well-defined vector/amplitude sets that provide the most neutral timbre", I don't know if I interpret vector/amplitude sets correctly but to define a sound field at the listener that provides the most neutral timbre, I guess one would need to have knowledge of how to go from "sound amplitude versus frequency versus angle of incidence versus time of incidence at the head" to perceived timbre, sounds like a very difficult problem to me.

Hi Victor,

I am not sure whether I should thank you for this “meaty” set of questions or not. I don’t think I can do them the justice they deserve in this particular setting.

My usual qualifier: The context is so large and complex, it is very difficult to convey meaningful answers by way of the reductionist short cut that is required by forum chatting. Each simplification will be misleading, but, I’ll provide information that hopefully at least points the way to understanding of the issues involved. Hopefully I won't have just created more confusion.

I’ll answer the first couple first and follow with more over the next few days.

Please excuse the non-standard formatting…

All the best,

- James

[[HOW DO YOU DEFINE SPATIAL NEUTRALITY?

THE SPATIAL ASPECTS OF REPRODUCTION (SENSATION OF ROOM, ENVELOPMENT, LIVELINESS ETC) OUGHT TO BE RATHER SUBJECT TO INDIVIDUAL TASTE, AND MOST RECORDINGS ARE NOT MEANT FOR "SOUND FIELD REPRODUCTION" BUT TO CREATE A PLEASURABLE EXPERIENCE...]]


I’ll address this in reverse:

While many people will have a subjective preferences regarding image size and form (such as those that like Bose 901’s that sometimes create a singer with a mouth that appears 2-meters wide), my goal is to replicate the apparent scale that one would perceive at a live event. And, yes, most recordings are not recorded very well spatially, but, I am attempting to recreate the live event as best I can with high quality recordings that capture the scale of a live acoustical event. One cannot consistently correct for a recording that is spatially distorted and I am not attempting to do so.

I don’t believe that spatial experience is strictly dictated only by preference and which would suggest that one shouldn’t try to make spatially accurate loudspeakers. I make designs as spatially accurate as possible for recordings that have accurate spatial cues, and if that is appreciated only by a small group of listeners then that is my risk.

In terms of how one defines spatial neutrality…

There are a number of accurate objective techniques, but here is a simple subject spatial calibration approach that most any ambitious person can try:
1) Attend a live acoustical event,
2) Listen carefully and create spatial memory models of total presentation size and individual instrument sizes
3) Capture the event with a well engineered recording
4) Repeat with multiple events of different scales including full orchestra in concert hall, small jazz trio in club, and in home, and single voice and guitar in one’s own sound room.

This will create a reference spatial template for further testing and calibration of loudspeakers systems
(Loudspeaker system includes all elements: Loudspeaker, environment, listener, electronics, etc)

With great effort, spatial reproduction can be accomplished that is surprisingly good and it doesn’t require exotic devices.

As far back as 1977 I worked with an avid classical concert fan that had reserved the same concert seat for many years (front row and center, first balcony). We had access to master recordings from that concert hall from live events that we attended. We attended performances and worked on his system for over a year until we had a perceived spatial impression at his listening seat in his home that closely matched that of his listening seat at the concert hall. A semi-near field use model was employed for a single listener.

With great effort and diligence this can be accomplished, but it requires rigor that very few people are willing to undertake. It could be done decades ago and it can be done now. It does not depend on the advancements in technology that we have today.

[[DO YOU MEAN THAT THE "REFERENCE SOUND FIELD" AT THE LISTENING POSITION DIFFERS WITH THE USE MODEL?]]

This is a very complex issue to convey.

There is more than one differentiated reference sound field that can all provide a lifelike facsimile of the original event.

A particular reference sound field may be more easily achieved by a given use model.

(Simply having a source height of the loudspeaker projecting at ear level or from below ear level as with (the clue) requires a different calibration relationship to define elevation field properly.)

This is the part that is not often well understood or at least not well quantified:
We don’t have to have the reproduced sound field around the head be identical to the original live sound field around the head to fool the ear/brain system into thinking it is hearing the original live event. Certain information can be altered or removed without an audible change. This is similar to perceptual coding used in transparent recording techniques. While much has been done in the area of digital compression, very little work has been done in quantifying this issue in psychoacoustics.

Two of the levels of investigation we have pursued into psychoacoustic perceptual calibration are;
1) what live event arrival data can be altered or removed before one can detect a JND (just noticeable difference) to the original tonal and spatial event;
2) what live event arrival data can be altered or removed wherein one may not hear the event exactly as they heard it in the original performance space but it is still heard as a live event (this would be similar to moving to a different seat from where the microphones are capturing the event, but still listening in the same concert hall, still perceiving the presentation to be a live event).
Fortunately, among all the information arriving at our ears in a live event, the ear/brain systems only requires a smaller subset of that information to determine the attributes of the live event. As we get a better understanding of that subset and what attributes can be ignored, we can simplify our information delivery requirement.

In the past the distortion of the live event signals delivered to the listener has been randomly based on limitations of our delivery system rather than a proactive choice of what characteristics can be removed or changed while maintaining the perceptual goal.

There is another variable that I will address next that is significant in the discussion.

More to come…

[Jacro]

9. Best sonic balance is achieved at a listening distance of between four and fourteen feet.

Thanks for a complete answer.
Is there no problem listening to a waveguide speaker at a distance of only 4 feet (1 meter)?
What is your opinion about the absolute ideal distance?

Hi Ragnwald,

I find that if the room dimensions are adequate, approximately 3-meters is quite ideal, but it also depends on the room width. If the loudspeakers are spaced with the proper width for the 3-meter distance, I would want to make sure they were not much closer than about 1-meter from the side walls.

As little as 1-meter listening is possible, as the crossover is calibrated to have the frequency response adapt to closer listening, which is similar to sitting in a higher chair or being taller, when listening at a greater distance (angles being similar relative to the ears from the driver array).

- James

[Jacro]

Geee, it sounds like you are dictating the 'user guide' of Ino/Guro speakers. Rather interesting how two persons sharing the same interest and, at least to some extent, sound philosophy can come up with the same user guidelines like 8000km apart, then again, maybe not!

Good Observation Glebster,

As you may know, SJÖFN HI FI, the current distributor of (the clue) was previously the U.S. importer and distributor of GURU. The owner of SJÖFN HI FI, Lars Erickson, was used to presenting similar installation instructions with GURU for two years or so.

As he started presenting (the clue) he continued with the familiar general presentation (with variations relative to (the clue)) , as the use-models are quite similar.

I will be preparing an Advanced User Guide in the near future to supplement the basic User Guidelines that currently come with (the clue).

So, not exactly an 8,000 km coincidence, but the instructions are certain to look very familiar in Sweden.

Cheers,

- James

[Jacro]

Hi Jacro,

First I want to say that it's nice to have you here.
I find your ideas interesting and the the way that you use the drivers in the 'clue' is
new to me. But there's allways some tradeoffs. So now I just want to ask, What do
you feel is the biggest sacrifice you have made when having a wideband driver?
How capabale is the clue when it comes to soundpressure?

Sometimes I just like to play it loud.

Rolf

Hi Rolf,

In terms of wideband drivers, there are a number of issues to deal with that have previously limited their performance. If I were going to make a single driver loudspeaker, I would take a very different approach than I did with (the clue). (the clue) uses a main driver that was conceived of as a full range driver, but the parameters were modified to match the dispersion driver, and since the system still uses a tweeter as a dispersion driver, the system isn't operating as a full-range driver system, as that type of system is normally defined. One might call it an 'enhanced' full-range driver system.

With wide-band drivers the first key is to minimize colorations due to diaphragm breakup at high frequencies. Besides a few other minor issues, the most significant remaining problems that need to be solved to create an effective device are mainly:

1) maintaining extended high frequency response while still having adequate linear excursion capability (X-max) is usually a conflict. Often, creating a long enough voice coil to achieve the desired excursion, will also create high frequency problems.

2) lastly and most importantly, a problem with most wide-band drivers is that their power response is poor, particularly at high frequencies, both in having limited dispersion, and uneven dispersion.

With (the clue) the system has an effective low frequency excursion capability while maintaining good high frequency behavior, and the nature of the dispersion driver matching the wide-band driver provides the desired power response.

In terms of sound pressure capability, the focus of (the clue) was on 'quality' more than 'quantity' of sound, but it is quite satisfying in terms of dynamic range and level capability when used in the room volumes it was designed for.

The owner of SJOFN HIFI (the distributor of (the clue))enjoys listening at high SPLs, so I had to be sure to satisfy his needs when developing (the clue).

But, I realize that it will not reach the levels in large rooms that some folks will require.

I hope this answers your questions...

All the best,

- James

[roren]

Yes, that answer my questions.Thank you, James.
It also makes me more interested.

Thanks again,

Rolf

[Jacro]

Hi Victor,

Below (at the bottom) is more discussion relative to your remaining questions.
Again, excuse the formatting. I copied your last questions and moved them to the end in [[CAPS]] for clarity, since the post is so complex. I hope it is clear.

I apologize ahead of time, if I have merely confused the issues even more.

I assume more dialog will be forth coming…

All the best,

- James

Hi James,

Since your answer is quite long I have made quite many quotes below:

Ultimately, one must arrive at a power response emitted from the loudspeaker system (the amplitude/response curve “emitted IN each direction”) that creates the “head related power response” (amplitude at each frequency “received FROM” each direction) that is appropriate to create timbral and spatial neutrality.

How do you define spatial neutrality? The spatial aspects of reproduction (sensation of room, envelopement, liveliness etc) ought to be rather subject to individual taste, and most recordings are not meant for "sound field reproduction" but to create a pleasurable experience...

There are well-defined vector/amplitude sets that provide the most neutral timbre, but they are inextricably related to the use configuration model of the loudspeaker device.

This includes the boundary relationship of the transducer, including 'global' listening environment boundaries and 'local' enclosure boundaries.

If the device is boundary coupled to the front wall, floor, sidewall, or free standing, requires a very different radiation pattern. And each of these placement models has a range of parameters to be defined, again related to the environment and enclosure relationships (front of enclosure to boundary, vs. back and sides of enclosure to boundaries).

Do you mean that the "reference sound field" at the listening position differs with the use model?
Of course, the loudspeaker design would differ according to the loudspeaker use model if the same reference sound field at the listening position is the goal.

And by "There are well-defined vector/amplitude sets that provide the most neutral timbre", I don't know if I interpret vector/amplitude sets correctly but to define a sound field at the listener that provides the most neutral timbre, I guess one would need to have knowledge of how to go from "sound amplitude versus frequency versus angle of incidence versus time of incidence at the head" to perceived timbre, sounds like a very difficult problem to me.

The use model also includes the relationship of the listener’s upper body to the transducer/enclosure, such as angle (vertical and horizontal), distance, etc.

Example: Depending on wave launch source position relationship to listener, the optimal power response will differ.

I guess it depends on what the goal of the source is, but if it is to generate the same perceived timbre regardless of source angle to the listener, then why should the design goal be different for different source angles? After all our brain should be expecting all modifications on the sound field that our body gives for different sound angles.

This is just a small subset of the variations that must be taken in to consideration.

Unfortunately, most loudspeaker manufacturers still don’t define a specific use model (because they want to let people use the speakers where ever it is convenient) so for most loudspeakers, it is not possible to accurately define an effective power response.

I agree that it is a pity that not more manufacturers define a "use model" for their loudspeakers. They will never have full control over how the speakers will sound in peoples homes.

Each use model, once clearly defined, has a unique optimal radiation pattern/power response.

Once the use model and boundary relationship and listener relationship is defined then a specialized set of measurements must be taken to calibrate the power response.

To be truly effective, this is not defined by just measuring the radiation pattern outward from the loudspeaker.

One approach is to measure the polar response outward from the device under test, with a more precision angle weighting function depending on angle, with certain angles tangent to the listening position differentiated from all other angles with a specific non-linear priority.

Then, with a varying boundary set (multiple room sizes and forms) one measures a “reception” power response at the listener’s torso (upper body).

Do you take actual measurements at a listening position with a listener present? How do you know what such measurements should look like?

Ultimately, while measuring the power response “IN all directions” from the loudspeaker, one is really only providing a beginning step in defining what is more important, which is what is the amplitude response of arrivals at the listener “FROM all directions”.

By having a defined set of amplitude vs. frequency arrivals “from each direction” one can work back to the loudspeaker device being calibrated for a power response of how it should radiate “in all directions”.

This is combined with a few other adjustments and design processes that impact the sound field around the listener’s head.

I think the most interesting problem to discuss here is what kind of sound field you want to have at the listening location - if you know that, then it's a separate problem how you should design loudspeakers for a certain use model to achieve this.

To make the matter even more complex, the ideal power response is not a single channel, emission definition, but must be calibrated to the channel count employed to achieve an optimal spatial response while maintaining timbral neutrality. This requires the power response from each loudspeaker to be recalibrated differently for “stereo” if timbre is to be maintained and “multi-channel coloration” minimized.

This is an interesting subject as well… Multiple loudspeakers will of course add in a complex manner. I imagine that they will add differently at a microphone than at a listener's head. And it should also depend on the music signal - if the sound is correlated or uncorrelated between the channels.

Then, the next step, depending on the architecture of the loudspeaker, ideally, one optimizes the polar response(s) to maintain neutrality across a listening window wide enough for at least three listeners seated beside each other. Some loudspeaker system topologies allow for this adaptation more than others.

Again, depending on the type of emission architecture (dipole, monopole, free-standing, ½ space, ¼ space..., etc.) and use model (listener/loudspeaker/environment relationship) the above stated calibration technique will result in a unique amplitude/vector set for each system type.

Hopefully, what I have written so far is of interest, even though it does not provide a quick and simple answer to the very important question that was asked.

I’ll see if there is feedback on what I’ve written so far, and if there is interest, we can explore further towards a more complete answer.

For my part there is a large interest in these questions. Thanks for the meaty post!

/Viktor

[[OF COURSE, THE LOUDSPEAKER DESIGN WOULD DIFFER ACCORDING TO THE LOUDSPEAKER USE MODEL IF THE SAME REFERENCE SOUND FIELD AT THE LISTENING POSITION IS THE GOAL.

AND BY "THERE ARE WELL-DEFINED VECTOR/AMPLITUDE SETS THAT PROVIDE THE MOST NEUTRAL TIMBRE", I DON'T KNOW IF I INTERPRET VECTOR/AMPLITUDE SETS CORRECTLY BUT TO DEFINE A SOUND FIELD AT THE LISTENER THAT PROVIDES THE MOST NEUTRAL TIMBRE, I GUESS ONE WOULD NEED TO HAVE KNOWLEDGE OF HOW TO GO FROM "SOUND AMPLITUDE VERSUS FREQUENCY VERSUS ANGLE OF INCIDENCE VERSUS TIME OF INCIDENCE AT THE HEAD" TO PERCEIVED TIMBRE, SOUNDS LIKE A VERY DIFFICULT PROBLEM TO ME.

I GUESS IT DEPENDS ON WHAT THE GOAL OF THE SOURCE IS, BUT IF IT IS TO GENERATE THE SAME PERCEIVED TIMBRE REGARDLESS OF SOURCE ANGLE TO THE LISTENER, THEN WHY SHOULD THE DESIGN GOAL BE DIFFERENT FOR DIFFERENT SOURCE ANGLES? AFTER ALL OUR BRAIN SHOULD BE EXPECTING ALL MODIFICATIONS ON THE SOUND FIELD THAT OUR BODY GIVES FOR DIFFERENT SOUND ANGLES.]]

One example would be that if the vertical source angle is a non-zero angle, then the ears will be receiving a different frequency response, due to angular pinnae interaction and also the vertical spatial location and size will be altered. So, for a non-zero vertical angle, the response from the loudspeaker may wish to be altered to simulate a zero vertical angle arrival.

Besides vertical angle, variation in source types such as point source or line source may need to each have a unique amplitude-vector to achieve a similar perceived sound field match and natural timbre perception.


[[I AGREE THAT IT IS A PITY THAT NOT MORE MANUFACTURERS DEFINE A "USE MODEL" FOR THEIR LOUDSPEAKERS.

THEY WILL NEVER HAVE FULL CONTROL OVER HOW THE SPEAKERS WILL SOUND IN PEOPLES HOMES.]]

Never say never. ☺


[[DO YOU TAKE ACTUAL MEASUREMENTS AT A LISTENING POSITION WITH A LISTENER PRESENT? HOW DO YOU KNOW WHAT SUCH MEASUREMENTS SHOULD LOOK LIKE?]]

Yes, we take measurements, and it is difficult to capture and quantify with conventional techniques. It requires an unusual, proprietary technique to achieve meaningful results.

Many folks tend to measure loudspeakers and sound fields with a single microphone. Most engineers with psycho-acoustical backgrounds recognize that one microphone doesn’t give us an accurate picture of what an ear-brain system senses in an acoustic space where a head is placed, because we have two ears.

But the truth is, that even a dual microphone arrangement or dummy head binaural system doesn’t begin to replicate the dual-ear/brain system. This is for a number of reasons, but a significant issue is that we don’t have just two ears.

When we listen to any sonic event, we don’t hold our heads still. Our two ears are constantly moving in space and a lot of how our hearing characterizes the spatial aspect (and to a degree tonal) is related to the movements of our head/ears in the space around our heads. Our ear brain system is constantly roaming “sampling” the three dimensional space, analyzing it, recalculating the information to create a dynamic, head related multi-dimensional transfer function.

If you clamp someone’s head and hold it in a fixed position, not allowing them to move, they loose part of their ability to accurately characterize a spatial event. Even though we don’t realize it, we move our heads all the time as we listen, tilting, rotating the angles of our ears on a micro basis.

This movement can be captured with accelerometers and translated to dynamic ear positions.

(This is a method that can be applied to headphones to create more accurate, out-of-the-head spatial development of reproduced program material.)

[[I THINK THE MOST INTERESTING PROBLEM TO DISCUSS HERE IS WHAT KIND OF SOUND FIELD YOU WANT TO HAVE AT THE LISTENING LOCATION - IF YOU KNOW THAT, THEN IT'S A SEPARATE PROBLEM HOW YOU SHOULD DESIGN LOUDSPEAKERS FOR A CERTAIN USE MODEL TO ACHIEVE THIS.]]

The relationship can be defined, but, as I have hopefully conveyed in the previous discussion, the multi-variable, dynamic, interactive nature of loudspeaker emission and near-head sound field requires a recursive method of matching the radiation of the loudspeaker to achieve the desired sound field at the listener and is not a simple, linear relationship. As much as it would be nice to provide a simplified representation, it would be misleading to do so.


[[THIS IS AN INTERESTING SUBJECT AS WELL… MULTIPLE LOUDSPEAKERS WILL OF COURSE ADD IN A COMPLEX MANNER. I IMAGINE THAT THEY WILL ADD DIFFERENTLY AT A MICROPHONE THAN AT A LISTENER'S HEAD. AND IT SHOULD ALSO DEPEND ON THE MUSIC SIGNAL - IF THE SOUND IS CORRELATED OR UNCORRELATED BETWEEN THE CHANNELS.]]

Yes, you are correct in that the summation is complex and has very little correlation between a single microphone and the ear-brain system. (As we discussed above, even monophonic transmission is received in a manner that is much more complex than any current microphone can interpret).

The application of two-channel stereo, of course must be integrated into the characterization from the beginning. It doesn’t work to start with optimization of mono channels and then just add two of them to have stereo. If one is to operate a two channel stereo-based system, the system should be developed as a two-channel system starting with the most basic elements of design.

The geometric relationship between a single pair of loudspeakers (passive, without signal processing) and the listener has an optimal use model that begins with, and is set by achieving the minimum inter-aural cross-correlation, which happens at approximately +/- 21 degree horizontal angles. At these angles there is an opportunity for correlated and uncorrelated signals being sorted out more effectively around the headspace. Use models, that don’t provide the +/-21 degree angle, are difficult to optimize without incorporating signal processing.

Correlated signals, that arrive correlated after room interaction transit, must have secondary arrivals, suppressed in level or be decorrelated by way of boundary interaction. Central front wall reflections, floor and ceiling reflections, all fall into this category. If these are not suppressed, spatial and tonal corruption will be difficult to overcome.

Ultimately, throughout all the issues being discussed here, it is important to keep in mind that angular arrivals don’t just impact spatial perception, but they impact tonal reception as well. So, as one makes any angle changes (vertical or horizontal) one must monitor both the spatial and spectral impact.

The stereo realm is by definition, one that doesn’t quite work from a mathematical transfer function standpoint, but by optimizing the system by way of a better understanding of how the ear-brain receives the signals, and fortunately, having the forgiveness of the powerful ability of our ear-brain system to adapt to the circumstances, we have the potential for a surprisingly good facsimile of the live event.

Again, I find these answers to probably be of little help and/or interest, as they don’t give rule of thumb, specific application information. But, I hope they provide a ground-work to establish at least partially, what some of the variables are and what must be dealt with before establishing a final calibration.

I will continue to attempt to provide useful answers to questions, but I am concerned it will be hard to satisfy, because, if there are too many questions in one post, or questions that are too complex, it is difficult to provide an answer that does the subject justice. On the other hand, if a question is simple and narrow, it most often has so many interactive variables, that it can’t be answered as simply as asked.

I always found that I couldn’t get my complete answers from some one giving a lecture or writing an article, but what they could provide, is a pointer, to help me see where to look to find my answers. If what I have to say here is at least thought provoking, in that manner, I’ll be satisfied.

That said, I’ll do my best to work towards answers that are useful and maybe over time in our discussion, we can arrive at a more complete understanding of how to achieve the best solutions.

Cheers,

- James

[celef]

If you clamp someone’s head and hold it in a fixed position, not allowing them to move, they loose part of their ability to accurately characterize a spatial event. Even though we don’t realize it, we move our heads all the time as we listen, tilting, rotating the angles of our ears on a micro basis.

This movement can be captured with accelerometers and translated to dynamic ear positions.

(This is a method that can be applied to headphones to create more accurate, out-of-the-head spatial development of reproduced program material.)

wow, that sounds supercool, have you tried such headphones?

[KarlXII]

If you clamp someone’s head and hold it in a fixed position, not allowing them to move, they loose part of their ability to accurately characterize a spatial event. Even though we don’t realize it, we move our heads all the time as we listen, tilting, rotating the angles of our ears on a micro basis.

This movement can be captured with accelerometers and translated to dynamic ear positions.

(This is a method that can be applied to headphones to create more accurate, out-of-the-head spatial development of reproduced program material.)

wow, that sounds supercool, have you tried such headphones?

http://www.dspeaker.com/en/products/headspeaker.shtml

[Jacro]

If you clamp someone’s head and hold it in a fixed position, not allowing them to move, they loose part of their ability to accurately characterize a spatial event. Even though we don’t realize it, we move our heads all the time as we listen, tilting, rotating the angles of our ears on a micro basis.

This movement can be captured with accelerometers and translated to dynamic ear positions.

(This is a method that can be applied to headphones to create more accurate, out-of-the-head spatial development of reproduced program material.)

wow, that sounds supercool, have you tried such headphones?

Celef,

Yes, I have. Being curious about such things, I was fortunate to have a brilliant engineer (Robert Williamson) that worked for me at Carver Corporation back in the 1990's who developed a prototype, but it required tremendous computing power relative to what was available at the time. It was quite impressive to listen to headphones with the sensing, and audio correction processing, as compared to standard, unprocessed headphones, particularly with binaural recordings.

As far as I know, the idea was originated by Mark Davis back in the early 1970's and he wrote his thesis at MIT on the topic. He is now with Dolby Labs.

While it was possible to realize working prototypes for the last couple decades, it has only recently become cost effective enough to produce consumer products based on the concept. Eventually, I think we will see all future headphones adopting this type of processing.

If you have the opportunity, be sure to listen to a set... you will be very impressed. It also provides a great reference tool for optimizing loudspeakers.

Cheers,

- James

[Jacro]

If you clamp someone’s head and hold it in a fixed position, not allowing them to move, they loose part of their ability to accurately characterize a spatial event. Even though we don’t realize it, we move our heads all the time as we listen, tilting, rotating the angles of our ears on a micro basis.

This movement can be captured with accelerometers and translated to dynamic ear positions.

(This is a method that can be applied to headphones to create more accurate, out-of-the-head spatial development of reproduced program material.)

wow, that sounds supercool, have you tried such headphones?

http://www.dspeaker.com/en/products/headspeaker.shtml

Hi KarlXII,

Thanks for identifying that source for the technology. I have written a review of their patent (US 7,672,809) for The Voice Coil Loudspeaker Journal, but I haven't had an opportunity to hear their product.

- James

[paa]

I believe that head movements are much more useful in a real acoustic event than in a 5.1 setup. In the 5.1 case it is much more likely that head movements tend to reveal the point sources of the rear speakers rather than to achieve a more exact sound space experience, especially if one does not have multiple rear speakers like in movie theatres.
Also these headphones, I do believe, only try to keep the sound field oriented towards the image, and are not useful for improving the ears ability to pick up exact directions of projected phantom sounds.

[luminous]

James,

It's really interesting to read your views of my qestions. I'll read it once more before I make any comments...

Best /Viktor

[luminous]

James,

You're right that it's a bit difficult to discuss these kinds of very involved subjects on a forum. But then I think the purpose of a forum discussion may not be to reach scientific-quality conclusions but to have a good entertaining discussion. And for me it is always rewarding to discover ways of thinking that I haven't considered before. I do admire people such as yourself and others who are willing to contribute their great knowledge in these discussions.

I understand your standpoint in defining spatial neutrality, I think it makes good sense. A spatial aspect of sound perception that I believe is very difficult to reproduce though using a stereo system is the room "ambience", the natural ambience of the listening room will add to the recorded ambience which cannot really be reproduced naturally from only two points in space in front of the listener. So that's where I think there is some room for different listener preferences in choosing the type of ambience and degree of liveliness of the listening room.

It seems to me that it would not be very profitable for most companies in the audio business to use their resources for researching the kind of intricate low-level subjects that you are talking about in this thread. How have you found time for this?

I really like your notion that only a subset of the information in the original sound field needs to be reproduced to yield a realistic experience, and that knowledge of this subset is really vital in designing the reproduction system. It is certainly obvious that a stero-recording can only store a small part of the information in the original sound field, but can still give a very pleasurable experience when reproduced over a good system.
This also explains that different use-models of loudspeakers may make different reference sound-fields at the listener position feasable.

Perhaps I came with too many questions at once to you. :) But I think there are more questions coming...

Best,
Viktor

[Jacro]

James,

You're right that it's a bit difficult to discuss these kinds of very involved subjects on a forum. But then I think the purpose of a forum discussion may not be to reach scientific-quality conclusions but to have a good entertaining discussion. And for me it is always rewarding to discover ways of thinking that I haven't considered before. I do admire people such as yourself and others who are willing to contribute their great knowledge in these discussions.
•••••••••••
I understand your standpoint in defining spatial neutrality, I think it makes good sense. A spatial aspect of sound perception that I believe is very difficult to reproduce though using a stereo system is the room "ambience", the natural ambience of the listening room will add to the recorded ambience which cannot really be reproduced naturally from only two points in space in front of the listener. So that's where I think there is some room for different listener preferences in choosing the type of ambience and degree of liveliness of the listening room.
•••••••••••
It seems to me that it would not be very profitable for most companies in the audio business to use their resources for researching the kind of intricate low-level subjects that you are talking about in this thread.
•••••••••••
How have you found time for this?
•••••••••••
I really like your notion that only a subset of the information in the original sound field needs to be reproduced to yield a realistic experience, and that knowledge of this subset is really vital in designing the reproduction system. It is certainly obvious that a stero-recording can only store a small part of the information in the original sound field, but can still give a very pleasurable experience when reproduced over a good system.
This also explains that different use-models of loudspeakers may make different reference sound-fields at the listener position feasable.
•••••••••••
Perhaps I came with too many questions at once to you. :) But I think there are more questions coming...

Best,
Viktor

Victor,

I agree, that the purpose of a forum like this is to entertain, be entertained, and have fun. I’m always trying to balance the tedious rigor of research with the fun that is the real reason we are all involved in this joyful endeavor of music and audio.

•••••••••••

You are correct with your concern about the room ambience issue.

I refer to it as the “Second Venue” problem. Whenever we have a larger “First Venue”, where the live performance takes place, and then attempt to reproduce that performance in a smaller “Second Venue”, the “Second Venue” dominates and overrides many of the ambience and spatial cues of the larger original venue.

This is a major issue, perhaps THE major remaining issue to resolve, (only equaled by the need for an effective, consistent recording capture standard).

Decades ago, when Acoustic Research did their famous live vs. recorded demonstrations they always did them outdoors, eliminating the venue differential problem.

We can solve the ‘Local Variables’ at the loudspeaker quite well, and have been able to do so for a long time, but the ‘Global Variables’ of the interactive environments and sound field around the head of the listener, are still problems that are generally left unresolved in the industry.

Most loudspeaker system types have the room as an enemy. One must find ways to have the loudspeaker either ignore the room (possible, but difficult with today’s technology) or to work with the room.

Loudspeakers that work with the room are usually boundary coupled, such as Carlsson, Larson, Ino-Audio, (the clue), Allison Acoustics, early Snell, early Audio-Pro, and very few others (a surprisingly small percentage of the total loudspeaker market). But, coupling to the first boundary is merely a first step toward the total solution required, and only a few loudspeaker purveyors design their products to resolve this issue and also very few provide any guidelines to allow the user to maximize their system to address the “Second Venue” issue.

This area is ripe with opportunity for manufacturer and end-user alike.

•••••••••••••••••••

First, I would have to respectfully disagree with you that it is not profitable for companies to use their resources to research these subjects, or to at least leverage the research that has been done by others. It is my belief that once these things are well understood, one can build much better products for much lower cost. While I enjoy attempting to advance the state of the art, it is even more exciting to be able to provide state of the art performance at a low price.

Ultimately, quality can be achieved in a cost effective manner if efficient principles are employed. One of my favorite concepts is the idea that the goal of any product should be that it is; “so advanced, that it is inexpensive”. When one understands the issues thoroughly, one can focus the expense only on the attributes that result in significant sonic improvements, and reduce cost in all other areas.

In terms of the products in the marketplace, I believe the problem is not a lack of technology. It is a problem of marketing choices/priorities and lack of understanding of what solutions are available today.

As the science fiction writer William Gibson has stated,
“The future is here, it just isn’t well distributed yet”.

•••••••••••••••••••••••

In terms of your question of how I have found time for this, I would have to say that I have been fortunate to have the freedom to pursue what I most enjoy, and get paid for it (The secret I never tell my employers, is that I would probably do it for free ☺ ).

I’ve done much research on my own, and I’ve had the good fortune to be well funded and provided with good facilities by different organizations over the years.

But, basically, I am too curious to stop working these things, whether I’m at work or during my spare time.

••••••••••••••••••••••

In terms of a “stereo” recording only storing a small part of the information in the original sound field, you are correct, but I think the standard methods of “stereo” recording is a vastly underutilized application of two-channels. Stereo is limited, but two-channel information storage does not have to be nearly as limited as current “stereo”.

I think there is an opportunity to standardize a more advanced form of two-channel spatial/timbral perceptual coding that could significantly advance the effectiveness of two-channel reproduction systems. Unfortunately, the generic stereo recording process is a rather haphazard practice that is substantially void of scientific guidelines.

If we merely utilized the state of current psychoacoustic knowledge to standardize the recording process, such that playback system designers had a form of Rosetta-stone codex to define playback system calibration, we could all be enjoying lifelike reproduction in our homes today.

Once again, to paraphrase, “The future is available to be here now, it just isn’t properly organized to be well distributed.”

•••••••••••

Don’t worry, more questions are welcome…as long as there is patience with my answers.

•••••••••••

All the best,

- James

[roren]

Jacro,

You have learnt me a lot about speakers. An other thing that I admire is the
way you handle all of our questions.
I have experienced this before, and it has been from guys like you with a practical knowledge.
To mention a few of them I have had this experience with Ingvar Öhman and a mr Robert C White. You are all guys with real knowledge.
Maybe this is what makes the different between a bussines man and a scientist.

Just wanted to say that.

Rolf

[Jacro]

Jacro,

You have learnt me a lot about speakers. An other thing that I admire is the
way you handle all of our questions.
I have experienced this before, and it has been from guys like you with a practical knowledge.
To mention a few of them I have had this experience with Ingvar Öhman and a mr Robert C White. You are all guys with real knowledge.
Maybe this is what makes the different between a bussines man and a scientist.

Just wanted to say that.

Rolf

Rolf,

Thank you for your very kind words.

It is an honor to be included in the group of Mr. White and Mr. Öhman, both of which have been very generous with their substantial knowledge of audio.

Warm regards,

- James

[paa]

Jacro,
Whith rooms and room boundries being so important for sound quality, what shape of listening room would you recommend, if one were free to build it new?
Would non-parallell walls be of a significant advantage, or are there other things more important to consider?

[Jacro]

I believe that head movements are much more useful in a real acoustic event than in a 5.1 setup. In the 5.1 case it is much more likely that head movements tend to reveal the point sources of the rear speakers rather than to achieve a more exact sound space experience, especially if one does not have multiple rear speakers like in movie theatres.
Also these headphones, I do believe, only try to keep the sound field oriented towards the image, and are not useful for improving the ears ability to pick up exact directions of projected phantom sounds.

Hi paa,

Yes, head movements allow the ear-brain to have increased ability to accurately analyze the acoustical aspects of a any sound field. That is better for improving perceptual accuracy in live events, such as mapping sound fields at musical concerts and accurately detecting the direction of a growling tiger.

But, as you surmise, the hearing enhancement from head movement actually makes our job more difficult in developing perceptually accurate sound reproduction systems, in that the greater hearing acuity derived from head movements helps the ear-brain systems catch us trying to deploy our sound field trickery, such as phantom images, and allows the ear brain to detect the actual sound sources (loudspeaker locations).

As an example, one of the tricks we can deploy is to change the spectral response to create height information from a loudspeaker. When the head is still, it depends on the frequency response above 4 kHz due to pinnae effects to determine sound source height. Our perceptual accuracy for identifying the height of a sound source is not very good compared to our excellent ability in accurately detecting tiny differences in horizontal sound sources. But, with head tilting, we can engage our horizontal source analysis ear-brain system in detecting height information more accurately, by tilting our head to the side, making one ear higher than the other, which gives us improved information about the height of an image, overriding the poorer performing spectral analysis of height information.

You are also correct to suggest that these head-movement sensing headphones are limited making stereo recordings sound more realistic. But, with pure, binaural recordings they can be very impressive.

Cheers,

- James

[Jacro]

Jacro,
Whith rooms and room boundries being so important for sound quality, what shape of listening room would you recommend, if one were free to build it new?
Would non-parallell walls be of a significant advantage, or are there other things more important to consider?

paa,

After trying many exotic room shapes over the years, a basic rectangular room form seems to be consistently the best and most predictable.

Strongly angled walls do not eliminate low frequency standing waves (just moves them in frequency in a fairly unpredictable manner) but slightly angled walls or paintings that are hung in a manner that are angled down slightly, centered at ear level can minimize higher frequency slap echo. Rooms with more than six sides can have unpredictable effects (good or bad) and peaked ceilings can have deleterious effects, sometimes causing focusing energy back at the listener, instead of diffusing the acoustic energy.

Generally, the larger the room the better, due to the fact that the first to second venue distortion is reduced and the transition to the modal region is moved to a lower frequency.

Of course there are other details relative to optimizing room acoustics, but in terms of your specific question, I would recommend the basic room shape to remain substantially six-sided and rectangular.

- James

PS - One exception is that of certain non-standard, non-rectangular room designs can be effective if the loudspeaker is truly part of the room, such as large waveguide or horn systems built into, and as part of the sound room, such that the room is an extension of the loudspeaker itself.

[paa]

I thought I was the first to invent it back in 1985, but it turns out that Bobby Beaver at Altec Lansing had developed products based on the concept at least 10 years earlier with the Model 814A “Extenda-Voice” and was granted a patent (US 3,722,616).

http://www.lansingheritage.org/html/alt ... rs/814.htm

Later work was done in 1982 by POLAR-PRO in Finland (WO8401681) and Skip Cross in America (US 4,437,541), and also in 2001 by Noselli in Italy, EP1137318A2.

Most recently, the technique can be seen in the Gradient Helsinki 1.5 loudspeaker, used as the midrange baffle:
http://www.gradient.fi/helsinki15/

Best regards,

- James

How is the width of such a device related to the cut off frequency?

[Jacro]

I thought I was the first to invent it back in 1985, but it turns out that Bobby Beaver at Altec Lansing had developed products based on the concept at least 10 years earlier with the Model 814A “Extenda-Voice” and was granted a patent (US 3,722,616).

http://www.lansingheritage.org/html/alt ... rs/814.htm

Later work was done in 1982 by POLAR-PRO in Finland (WO8401681) and Skip Cross in America (US 4,437,541), and also in 2001 by Noselli in Italy, EP1137318A2.

Most recently, the technique can be seen in the Gradient Helsinki 1.5 loudspeaker, used as the midrange baffle:
http://www.gradient.fi/helsinki15/

Best regards,

- James

How is the width of such a device related to the cut off frequency?

paa,

In its basic form, approximately 1/2-wavelength, similar to a dipolar device.

But, since it has the rear waveguide as an additional variable, one can vary the front-to-back output ratio, allowing significant modification of directivity and also changing cut-off frequency to some degree.

- James

[paa]

Thanks for the explanation.
But it seems strange that the Gradient Helsinki 1.5 can have as low crossover frequency as 200 Hz between the cardioid mid and the bass, considering the mid housing only seems to be approximately 300 mm in diameter.
Can there be any other trick in there, to lower the cut off frequency?

[Jacro]

Thanks for the explanation.
But it seems strange that the Gradient Helsinki 1.5 can have as low crossover frequency as 200 Hz between the cardioid mid and the bass, considering the mid housing only seems to be approximately 300 mm in diameter.
Can there be any other trick in there, to lower the cut off frequency?

Yes, there are a number of "tricks", and Jorma Salmi, the developer of the Helsinki, is a very capable loudspeaker engineer.

Steps to extend Di-Cardioid low frequency cut-off:

1) The output through the rear waveguide is greater than the axial energy from the front of the device, so it is most often best to use damping material in rear waveguide to equalize the front and rear amplitudes. By doing so, the effective internal transit length of the waveguide will increased such that the 'virtual' dimension of the waveguide will be effectively increased by a small amount.

2) By setting the resonant frequency of the transducer at a predetermined frequency below the cut-off frequency of the waveguide, and adapting the motor strength, suspension compliance and cone mass to provide an under-damped transducer 'Q' of between 1.1 and 2.0 at the resonant frequency, one can extend the LF response of the driver/waveguide combination by about at least a half-octave, a full octave in some cases. (This approach is used with many dipole systems to extend their response below baffle cut-off.)

3) At or near the crossover frequency, one can under-damp the high-pass filter that drives the waveguide transducer creating some narrow-band gain in exchange for a dip in the impedance curve. If one observes the separate amplitude responses of the woofer and the midrange and the impedance curve of the Gradient Helsinki loudspeaker, it will be seen that there is an impedance minimum at 250 Hz and a corresponding peak (about 2 to 3 dB) in the amplitude response of the high-pass section at the same frequency, suggesting the use of an underdamped high-pass filter, boosting the waveguide response enough to achieve an acoustic crossover at 230 Hz.

By applying these techniques, it one can see how the Helsinki achieves the extension of its 575 Hz baffle cutoff frequency down to the 230 Hz crossover frequency.

There are other subtle aspects that are incorporated into the Helsinki to smooth the midrange response of the Di-Cardioid waveguide.

A centrally mounted transducer on a circular Di-Cardioid waveguide will have ripple in the on-axis response at each half-wavelength multiple. By mounting the midrange transducer off-center in the waveguide, and tilting the waveguide back, such that it is pointing upward somewhat, the ripple in the response is distributed more evenly, and the waveguide still operates down to substantially the same cut-off frequency.

I hope this adds some clarification about how to use these types of devices.

Cheers,
- James

[Kraniet]

I think the topic has been dealt with to some extent in the thread. But I still would like to hear a little more about your views on the shape of the frequency curve of the speakers. Its something that have been discussed quite alot on this forum. Mr Öhman calls it "stereo-system compensation", sayin that the "psychoacoustical correct frequency response isnt the "straight line" that so often is discussed.

Youve mentioned earlier that most companies just do an arbitrary "BBC-dip".

But Toole and harman (Revel) say that the curve should be straight as an arrow with an even fall in dispersion towards high frequencies. Is this just because their evaluation process is mono only?

One example is



Being a schematic picture it lacks any dB-reference. But the general idea is visible.

What are your views on this? Do you have a particular "target curve" that you aim for? Do "the clue" have any "compensations" for a better resolution in regards toward the "stereo-system fault"?

Are there any measured curves out (of the clue) on the web that you know of or could you even provide some yourself?

[Jacro]

I think the topic has been dealt with to some extent in the thread. But I still would like to hear a little more about your views on the shape of the frequency curve of the speakers. Its something that have been discussed quite alot on this forum. Mr Öhman calls it "stereo-system compensation", sayin that the "psychoacoustical correct frequency response isnt the "straight line" that so often is discussed.

Youve mentioned earlier that most companies just do an arbitrary "BBC-dip".

But Toole and harman (Revel) say that the curve should be straight as an arrow with an even fall in dispersion towards high frequencies. Is this just because their evaluation process is mono only?

One example is



Being a schematic picture it lacks any dB-reference. But the general idea is visible.

What are your views on this? Do you have a particular "target curve" that you aim for? Do "the clue" have any "compensations" for a better resolution in regards toward the "stereo-system fault"?

Are there any measured curves out (of the clue) on the web that you know of or could you even provide some yourself?

Kraniet,

This is an interesting topic. There are many different frequency ranges that can be purposefully adapted and they all interact with each other, so once again, analyzing and applying any correction factors in isolation from other interactive parameters has limited usefulness. But, for the purposes of this discussion, we’ll limit ourselves to elements relating to the band of frequencies that relate to spectral shifts due to two channel cross-talk compensation.

First, I do have a target curve, but it is a three dimensional set of power response curves that also include other interactive variables. And, ( the clue ) does incorporate a specialized response to address the "stereo-effect".

I believe that Toole and Voeks remain fixed on a simplified response characteristic, partly, due to the fact that they appear to want to sustain a non-specific, generalized use model that a large company like Harman maintains so as to not limit customer interest. I believe Toole has provided a good service to the audio industry, but it is aimed more towards how to optimize standard configuration loudspeakers applied in the most general arrangements.

If one participates in their loudspeaker comparison test room, it becomes obvious that their approach is biased towards standard free standing loudspeakers, and a boundary coupled (or many other innovative types of systems) cannot be effectively evaluated within their test setup). It is unfortunate, in that they have tremendous resources, but their focus in on how to optimize standard devices, not how to create a new, state of the art device that would involve any new level of packaging, room coupling, and radiation pattern novelty.

While specialized frequency responses are useful, they can only be advantageous if a specific use-model is deployed. Otherwise, any modification to the frequency response will be heard as merely an undesirable coloration.

That said, while the phantom image creation works well with most listeners (not all listeners hear phantom images) the “stereo-effect” has a number of well known, undesirable elements. The unintended crosstalk of the sound from the right loudspeaker arriving at the left ear and mixing with the left loudspeaker output causes both a spatial limiting effect and a spectral comb filtering that form linear distortions of the original signal.

This can either be ignored or addressed, but to address it requires a balanced, organic approach with all aspects interacting to achieve a correction without causing further perceptual errors. Some of what I do is proprietary, but I am glad to share some of the basic concepts that are involved in providing a corrective frequency response.

A simplified portion of one solution is, first, to optimize the listening angles, and polar amplitude responses to achieve the minimal inter-aural cross-correlation (IACC). With the loudspeaker/listener referenced for optimal IACC minimization, then the second step is to observe the remaining cross-correlated mixing and to apply a correction curve.

If one attempts this correction curve without first minimizing IACC by way of establishing loudspeaker/listener angles, the correction curve will have to be too severe, and tonal coloration from the correction will be heard.

If one takes a single sound source and move it around the head, the balance of minimized IACC without introducing too much spectral distortion, happens at about 21 degrees.

Once this is established, then one can introduce a correction curve of partial inverse ripple to reach a perceptually neutral balance.

By partial, I’m referring to both the amplitude of the ripple and the number of center frequencies in the series that one addresses.

I’ll give a little specificity here; I address the first three frequencies of the series with a weighting function that is descending as the amplitude is adjusted from the fundamental, to the second and third ripple harmonic (barely touching the third). Any harmonics above the third tend to be variable due to environmental and human interface issues and also, at those frequencies, one is starting to influence the height perception pinnae response, which is a separate, but interrelated, issue.

This is a one dimensional view of the issue, in that as one applies these types of corrections, one has to account for other influential variables depending on radiation pattern type and change with frequency, and multi-driver interactions and their influence at different angles, etc.

One aspect is that the induced correction ideally varies with angle, such that the off axis response ideally has the corrective curve fade out, in that one doe not want the reverberant response to include a dominant correction curve.

This isn’t a comprehensive answer (impossible without the context of a given set of variables), but hopefully one that shows a direction to explore for those who are interested in doing so.

Each optimization of this type may seem subtle to some listeners, but as one combines all of the techniques, to achieve a systems approach, they result in an organic whole that provides a musical perception that can be quite special, elevated from the standard, but “optimized for general use” loudspeakers that are derived from standard practice.

These approaches are particularly advantageous with boundary-coupled loudspeakers and/or loudspeakers that also minimize early reflections.

First, these types of systems can remove many of the extraneous acoustical arrivals.

Secondly, some claim that a boundary-coupled loudspeaker has reduced perception of front to back depth. While, that can be true for some boundary-coupled designs that don’t incorporate the full potential of the design approach, a well-designed directivity controlled, boundary coupled device will create a sound field that has greater front to back depth, by moving the frontal portion of the image forward. This is a natural effect of minimizing IACC and also reducing early reflections.

In general, I’d suggest that one pursue the conventional approaches, because the diligence and rigor it takes to balance all the parameters, once one attempts to truly optimize for all the system variables, such as the weakness of stereo, is rather tricky.

But, if one is willing to be adventurous, and diligent, the potential is for a sonic presentation that is very special.


Cheers,

- James

[Kraniet]

very interesting and thank you for finding time to answer these questions.
I assume that these corrections can only be evaluated properly in a stereo setup?

Im currently in the process of constructing a crossover myself. And I find that corrections of one or two dB arent noticeable when listening to just the one speaker. Next step is to buy more parts and do the optimisation on the stereo pair.


Would you say its best to start from a prefectly flat frequency response and introduce the different "tweaks". Or could a start with a rough "BBC-template"? If so how would this shelf look like?
I realiaze you cant disclose specifics but any pointer would be welcome.

Would you say that some (or all) of these compensations relate to "personal taste" or would you call them "universally true"?

By the way its a wall hung speaker with a angled baffle to get those (approx)22 degrees that Im building. Im gonna use it with a 10cm thick absorber on the inner side of the speaker. But I still get a pretty substantial dip at 350Hz even with the absorber.
What are your thoughts on this dip (I assume the clue get a similar dip) and how did you deal with it?

Heres a imigae of the measurement.(taken, laying on the ground, outside with a good distance to other objects)

[Nattlorden]

In 1979 Robert Carver created a modified version of the JVC unit and offered a processor recalibrated for stereo recordings called Sonic Holography.

All were attempts to fix some of the fundamental spatial and tonal flaws of 2-channel stereo. Theoretically, under ideal conditions, it can be a superior approach, but is much more critical to optimize than conventional stereo. While potentially better than stereo, if it isn’t perfectly calibrated it can sound much worse than stereo. Ultimately, it is best suited for reproducing a binaural based recording.

These types of systems were always problematic with widely spaced loudspeakers, because the cancellation signals were very difficult to match, as they had to include the frequency response effects of the sound rapping around the face, to the opposite ear.

I'm actually going to give this a go... Have one of the later Carver processors on it's way by mail... just for the heck of having tried that too. Fun thing that you mentioned it here. My curiousity was actually sparked by listening to an interview with Bob... and being one thing that I haven't tried before, I felt I needed to give it a try even if I'm well aware that it most likely turn out to be a waste of money.

[Jacro]

very interesting and thank you for finding time to answer these questions.
I assume that these corrections can only be evaluated properly in a stereo setup?

Im currently in the process of constructing a crossover myself. And I find that corrections of one or two dB arent noticeable when listening to just the one speaker. Next step is to buy more parts and do the optimisation on the stereo pair.


Would you say its best to start from a prefectly flat frequency response and introduce the different "tweaks". Or could a start with a rough "BBC-template"? If so how would this shelf look like?
I realiaze you cant disclose specifics but any pointer would be welcome.

Would you say that some (or all) of these compensations relate to "personal taste" or would you call them "universally true"?

By the way its a wall hung speaker with a angled baffle to get those (approx)22 degrees that Im building. Im gonna use it with a 10cm thick absorber on the inner side of the speaker. But I still get a pretty substantial dip at 350Hz even with the absorber.
What are your thoughts on this dip (I assume the clue get a similar dip) and how did you deal with it?

Heres a imigae of the measurement.(taken, laying on the ground, outside with a good distance to other objects)

Kraniet,

You are welcome.

Many loudspeaker colorations can be evaluated on a mono basis, and even within the current topic of tonal correction for stereo cross-talk linear distortion” much can be done by listening to on a monophonic, single loudspeaker basis (preferably mounted with the same spatial relationship angle to the listener as the final use model) or dual loudspeakers with mono program, to evaluate tonal aberrations independent of spatial effects, but the final evaluation must be done in the actual, use model, dual loudspeaker mode.

Generally I recommend starting from a flat axial and well behaved off-axis response as a first step, to isolate any system resonances, driver interactive effects, diffraction issues, or other a problems before optimizing the system power response. There are many exceptions, wherein the transducers are purposely designed to have a built in ‘effect curve’.

The goal of an compensations I am discussing here are an attempt to address universal objective issues, not for the purpose of subjective personal taste enhancements.

If one were to consider adjust a loudspeaker to personal taste, I would recommend getting everything objectively correct first, so that any preference distortions that one adds are easily adjusted and identifiable, instead of just being a pleasant artifact/distortion of the loudspeaker that is left in tact. This way, one can control the system more easily, having full understanding of what is being added, with the ability to get back to an objective baseline.

If you are seeing this dip single ground plane, outdoor measurements, it may be misleading. A single boundary reflection is often difficult to eliminate completely, but the room has many boundaries, and if the loudspeaker is purposefully placed for an optimal ratio front wall, floor, and sidewall distances (and ideally, ceiling distance also), then the boundary reflections will average out in a manner that will substantially eliminate any severe dips as you observed in your outdoor measurement.

Applying absorbent to the front and sidewalls will help further, as will some form of absorbent or diffusor on the floor. All of these tactics used together will tend to smooth the ripple to an acceptable level.

Only having a single reflection boundary is rather artificial. In fact, in some room situations, if you were to eliminate all the reflections 100%, except having one strong reflection remaining, it would sound worse than if you had all the reflections distributed in an effective manner, such as establishing golden ratio (~1.6) secondary path-length ratios.

Correlated reflections are psycho-acoustically the most disturbing (those arriving equally to both ears, such as floor reflection, front wall reflection, and ceiling reflections) particularly if just one dominates.

Make sense?

- James

[IngOehman]

I think the topic has been dealt with to some extent in the thread. But I still would like to hear a little more about your views on the shape of the frequency curve of the speakers. Its something that have been discussed quite alot on this forum. Mr Öhman calls it "stereo-system compensation", sayin that the "psychoacoustical correct frequency response isnt the "straight line" that so often is discussed.

Youve mentioned earlier that most companies just do an arbitrary "BBC-dip".

But Toole and harman (Revel) say that the curve should be straight as an arrow with an even fall in dispersion towards high frequencies. Is this just because their evaluation process is mono only?

One example is



Being a schematic picture it lacks any dB-reference. But the general idea is visible.

What are your views on this? Do you have a particular "target curve" that you aim for? Do "the clue" have any "compensations" for a better resolution in regards toward the "stereo-system fault"?

Are there any measured curves out (of the clue) on the web that you know of or could you even provide some yourself?

Hi everybody!

Just for the protocol, I'd like to clarify that all the problems which I include
in what I like to call "the stereo system flaws" or "the intrinsic flaws of the
stereo system", are problems in MANY different domains.

I.e. both the timbral domain, imaging distortions, dynamic (and pseudo
dynamic (linear behaviours that still reduce dynamic range)) effects and
also time resolution loss effects.

I say this, since I sense in the above quoted, that Kraniet has reduced it
all, or close to everything to being a question of frequency response - and
even to the idea that a single univocal target curve could be "the answer to
what constitutes as a correct compensation".

Nothing can be further from the truth.

All these things are delicately entangled in quite an illusive and convoluted
manner...


I do not like simplifications, and I really do not like to have things I've said
simplified by others - and then still being hold responsible!

I'm not responsible.

(Actually - I'm one of the most irresponsible normalized earthlings that I
know. I am to be trusted about everything - but with nothing.)

- - -

The problems also occur both during "encoding" and "decoding" (recording
and replaying the recording), both are exhibiting differens set of problems,
often cross-depending in intricate patterns, which actually creates quite a
few interesting possibilities to optimize all balances.

Anyway; the possibilities to find and optimize "the stereo system compen-
sations" are no less multifaceted than the originating problem is. And thus
the compensations (optimally applied) are also addressing the behaviours
in ALL of the mentioned domains.

- - -

For practical reasons, I try to avoid going into detailed descriptions of the
problems and solutions on internet foras.

BUT - I believe that I have been very clear (also here on faktiskt.se) that
even if you ONLY look on the timbral problems (ignore ALL the problems
that are manifesting in the other domains) caused by the stereo system
(i.e. by trying to pack the complexity of real life multi dimensional sound
into only two one-dimensional channels) and also ignoring* all radiating
directions other than the one aiming directly at the lister - I still do divide
both the problem and the sollution in 12 differens components - resulting
in quite different 0 degree frequency responce curves, depending on all
other parameters of the loudspeaker.

So there are no single target curve, and I do not see how there can even
be one! The idea of a single univocal target curve goes against everything
I know about the inner workings of our hearing.

What can be, and I believe is - is a complex equation that can be used to
create a target curve for one specific loudspeaker (assuming the environ-
ment is reasonable predictable).

I use such an equation, but again - there are at least 12 different aspects
of it where 11 are variables! (I include the 12 important enough to play a
role that is larger than the uncertainties.)

And again - the timbral part of the stereo system flaws are only a minor
part of the everything that is caused by the stereo system.

- - -

I just wanted so say that, so that no one attributes the idea to me, of the
stereo system flaws possible corrections being nothing but a target curve
in the timbral domain.

This said, I hope that no one misunderstands what I just wrote to be an
attach aimed against the BBC-dip, or the men behind it. To the contrary,
to my knowledge, the BBC's studies ware amongst the first in the word
addressing problems of the stereo system itself, and though being just a
fraction of a fraction of the truth, it does not diminish the fact that a first
step is often the most important - since it leades the way by pointing out
the direction.


Best regards, Ingvar

- - - - -

PS. Please excuse my English. Writing in English make me feel like my
head is full of brake fluid. Well, not that I've tried that...


*When I spoke about ignoring other radiation directions, I did not really
mean ignoring it (it is a vital part of the equation) only ignoring having it
adjusted separately - as a simplification, to point out that it is still very
complicated, even after such a simplification.

In real life however, engineering loudspeakers, I do no such simplifications,
but try to control everything and give each parameter the properties I like
it to have to work well in the application.

Not just "see what I got" and then try to do the best of it in regards of the
stereo system corrections.

[IngOehman]

PPS. I do not agree that using +/- 21 degrees (minimum IACC-points)
minimizes the intrinsic flaws of the stereo system. Actually, it depens on
the relation between the listening distance, the directivity index of the
loudspeakers and the acoustic radius of the room - in my opinion/from
my experience.

I suggest a somewhat wider opening angle of +/- 23 degrees for front
row listeners.

Two interesting positiv bi-effects from such a set up, is:

1. That the problem of some of the stereo system flaws, peaking at +/-
19-20 degrees, are counteracted/compensated by the higher degree of
reflexted sound further back in the listening room.

2. That the holofonic soundstage will be less widht-modulated by the
change of distance to the soundstage opening (the loudspeaker line), i.e.
creating a much more stable staging, wich corresponds much better to a
true sound source in "the room behind the loudspeakers".

3. That compatibility to the silver screen standards (+/- 22,5 degrees in
directors seat) are better.

[IngOehman]

PPPS. Sorry for the kidnapping of this thread.

I just ment to do one posting, to clear up what was said regarding a target
curve - mentioned together with my name, that I by accident happened to
see. But NOW I'm done!


/iö

[Jacro]

I think the topic has been dealt with to some extent in the thread. But I still would like to hear a little more about your views on the shape of the frequency curve of the speakers. Its something that have been discussed quite alot on this forum. Mr Öhman calls it "stereo-system compensation", sayin that the "psychoacoustical correct frequency response isnt the "straight line" that so often is discussed.

Youve mentioned earlier that most companies just do an arbitrary "BBC-dip".

But Toole and harman (Revel) say that the curve should be straight as an arrow with an even fall in dispersion towards high frequencies. Is this just because their evaluation process is mono only?

One example is



Being a schematic picture it lacks any dB-reference. But the general idea is visible.

What are your views on this? Do you have a particular "target curve" that you aim for? Do "the clue" have any "compensations" for a better resolution in regards toward the "stereo-system fault"?

Are there any measured curves out (of the clue) on the web that you know of or could you even provide some yourself?

Hi everybody!

Just for the protocol, I'd like to clarify that all the problems which I include
in what I like to call "the stereo system flaws" or "the intrinsic flaws of the
stereo system", are problems in MANY different domains.

I.e. both the timbral domain, imaging distortions, dynamic (and pseudo
dynamic (linear behaviours that still reduce dynamic range)) effects and
also time resolution loss effects.

I say this, since I sense in the above quoted, that Kraniet has reduced it
all, or close to everything to being a question of frequency response - and
even to the idea that a single univocal target curve could be "the answer to
what constitutes as a correct compensation".

Nothing can be further from the truth.

All these things are delicately entangled in quite an illusive and convoluted
manner...


I do not like simplifications, and I really do not like to have things I've said
simplified by others - and then still being hold responsible!

I'm not responsible.

(Actually - I'm one of the most irresponsible normalized earthlings that I
know. I am to be trusted about everything - but with nothing.)

- - -

The problems also occur both during "encoding" and "decoding" (recording
and replaying the recording), both are exhibiting differens set of problems,
often cross-depending in intricate patterns, which actually creates quite a
few interesting possibilities to optimize all balances.

Anyway; the possibilities to find and optimize "the stereo system compen-
sations" are no less multifaceted than the originating problem is. And thus
the compensations (optimally applied) are also addressing the behaviours
in ALL of the mentioned domains.

- - -

For practical reasons, I try to avoid going into detailed descriptions of the
problems and solutions on internet foras.

BUT - I believe that I have been very clear (also here on faktiskt.se) that
even if you ONLY look on the timbral problems (ignore ALL the problems
that are manifesting in the other domains) caused by the stereo system
(i.e. by trying to pack the complexity of real life multi dimensional sound
into only two one-dimensional channels) and also ignoring* all radiating
directions other than the one aiming directly at the lister - I still do divide
both the problem and the sollution in 12 differens components - resulting
in quite different 0 degree frequency responce curves, depending on all
other parameters of the loudspeaker.

So there are no single target curve, and I do not see how there can even
be one! The idea of a single univocal target curve goes against everything
I know about the inner workings of our hearing.

What can be, and I believe is - is a complex equation that can be used to
create a target curve for one specific loudspeaker (assuming the environ-
ment is reasonable predictable).

I use such an equation, but again - there are at least 12 different aspects
of it where 11 are variables! (I include the 12 important enough to play a
role that is larger than the uncertainties.)

And again - the timbral part of the stereo system flaws are only a minor
part of the everything that is caused by the stereo system.

- - -

I just wanted so say that, so that no one attributes the idea to me, of the
stereo system flaws possible corrections being nothing but a target curve
in the timbral domain.

This said, I hope that no one misunderstands what I just wrote to be an
attach aimed against the BBC-dip, or the men behind it. To the contrary,
to my knowledge, the BBC's studies ware amongst the first in the word
addressing problems of the stereo system itself, and though being just a
fraction of a fraction of the truth, it does not diminish the fact that a first
step is often the most important - since it leades the way by pointing out
the direction.


Best regards, Ingvar

- - - - -

PS. Please excuse my English. Writing in English make me feel like my
head is full of brake fluid. Well, not that I've tried that...


*When I spoke about ignoring other radiation directions, I did not really
mean ignoring it (it is a vital part of the equation) only ignoring having it
adjusted separately - as a simplification, to point out that it is still very
complicated, even after such a simplification.

In real life however, engineering loudspeakers, I do no such simplifications,
but try to control everything and give each parameter the properties I like
it to have to work well in the application.

Not just "see what I got" and then try to do the best of it in regards of the
stereo system corrections.

Hello Ingvar,

I both agree and empathize with you regarding the concern and danger of having ones forum comments reduced to oversimplifications.

The multi-dimensional aspects of effective endeavors in loudspeaker engineering are most often far more complex that what can be reduced to a few paragraphs, but, we are a social animals with a shared passion, so we embark on a dialog about the things we care about.

It seems that maybe the best one can do in this type of forum is to entertain, bring some joy and to provide incomplete ideas that hopefully at least point in a direction that empowers others to be a little more effective at discovering their own answers….and in the mean time, hope that one’s statements aren’t terribly misconstrued as they are passed on and repeated.

Anyhow, for my part, I appreciate your use of English and tolerating the sensation of a brake fluid filled head.

All the best,

- James

[single_malt]

I do not like simplifications, and I really do not like to have things I've said
simplified by others - and then still being hold responsible!

I'm not responsible.

Yes you are. By writing lots of incoherent portions of information often concluded with sentences like "at least 12 different aspects"(without further explanation) and "All these things are delicately entangled in quite an illusive and convoluted manner... "
No wonder people start tryning to make their own interpretations to get the whole picture understandable..

[UrSv]

If anybody is incoherent it is you. Again and as always. Could you please stop bringing your personal agenda och stalking IngOehman out of this thread please? We have a competent and sharing international contributor here and it would be nice of you not to soil this thread with your ever repeating attacks on IngOehman. Grow up.

[stefanolo]

I do not like simplifications, and I really do not like to have things I've said
simplified by others - and then still being hold responsible!

I'm not responsible.

Yes you are. By writing lots of incoherent portions of information often concluded with sentences like "at least 12 different aspects"(without further explanation) and "All these things are delicately entangled in quite an illusive and convoluted manner... "
No wonder people start tryning to make their own interpretations to get the whole picture understandable..

Here we (you) go again ...speaking of predictable and provocative manners....As you might have noted Jacro, and I figure a lot of others (not occupied by harassing IngOehman whenever the possibility occurs), sympathize with Ingvar in not allways explaining every aspect of a technical issue by explaining every atom involved.

I hope IÖ doesn't respond to your provocation and this thread goes down the drain.


Sorry for OT

/stefan

PS. Jacro, I very much appreciate you beeing around and giving your views on these very complicated matters even though I'm not technically educated at all. Thanks!

[Kraniet]

Kraniet,

You are welcome.

Many loudspeaker colorations can be evaluated on a mono basis, and even within the current topic of tonal correction for stereo cross-talk linear distortion” much can be done by listening to on a monophonic, single loudspeaker basis (preferably mounted with the same spatial relationship angle to the listener as the final use model) or dual loudspeakers with mono program, to evaluate tonal aberrations independent of spatial effects, but the final evaluation must be done in the actual, use model, dual loudspeaker mode.

Generally I recommend starting from a flat axial and well behaved off-axis response as a first step, to isolate any system resonances, driver interactive effects, diffraction issues, or other a problems before optimizing the system power response. There are many exceptions, wherein the transducers are purposely designed to have a built in ‘effect curve’.

The goal of an compensations I am discussing here are an attempt to address universal objective issues, not for the purpose of subjective personal taste enhancements.

If one were to consider adjust a loudspeaker to personal taste, I would recommend getting everything objectively correct first, so that any preference distortions that one adds are easily adjusted and identifiable, instead of just being a pleasant artifact/distortion of the loudspeaker that is left in tact. This way, one can control the system more easily, having full understanding of what is being added, with the ability to get back to an objective baseline.

If you are seeing this dip single ground plane, outdoor measurements, it may be misleading. A single boundary reflection is often difficult to eliminate completely, but the room has many boundaries, and if the loudspeaker is purposefully placed for an optimal ratio front wall, floor, and sidewall distances (and ideally, ceiling distance also), then the boundary reflections will average out in a manner that will substantially eliminate any severe dips as you observed in your outdoor measurement.

Applying absorbent to the front and sidewalls will help further, as will some form of absorbent or diffusor on the floor. All of these tactics used together will tend to smooth the ripple to an acceptable level.

Only having a single reflection boundary is rather artificial. In fact, in some room situations, if you were to eliminate all the reflections 100%, except having one strong reflection remaining, it would sound worse than if you had all the reflections distributed in an effective manner, such as establishing golden ratio (~1.6) secondary path-length ratios.

Correlated reflections are psycho-acoustically the most disturbing (those arriving equally to both ears, such as floor reflection, front wall reflection, and ceiling reflections) particularly if just one dominates.

Make sense?

- James

Yes it makes sense. One can really understand how the "established" HiFi-brands arent into these kind of compensations.
But I find it to be a pity aswell. Feels I bit of a shame that companies like B&W or Revel spend all that money on designing new high-tech transducers and put them in advanced boxes with "high-end" expensive crossover parts only to have a sub-optimal crossover that doesnt make the best of the stereoflaw.
On the other hand those companies seem to have the best ability to incorporate the right compensations. Would you say the concept of the stereoflaw is to difficult a matter for "normal" people(even HiFi-nuts)? As always I assume its the salespeople that dictates what can be done.

Regarding the compensation curve I posted above Im avare that it isnt as simple as that. And I apologize if thats how it seemed. I just brought it in as an example of how it could look like. Being a generalist I tend to speak in general terms even when I know it isnt as easy as that. :)

I guess these compensations are percieved differently if its a big speaker or a small. Different placements of the speaker in the room would also affect this. So a firm advice on placement should be a minimal when it comes to loudspeakers, wonder why the industry are so afraid of that..

What are your feelings when it comes to multichannel sound? Are these compensations still good to have or should they look different (not exist even)?
What are your views on multi channel sound in general. Reading Tooles book I got the expression that he seem to think multi channel are the way to go forward? Seeing how difficult it is to make a stereo pair work in an optimal way, it feels daunting to have 5 or more speakers with the "right" compensations applied. Or maybe it isnt all that critical?
The lack of an stardard when mixing surround sound it might be hard to generalize?

[Jacro]

Kraniet,

You are welcome.

Many loudspeaker colorations can be evaluated on a mono basis, and even within the current topic of tonal correction for stereo cross-talk linear distortion” much can be done by listening to on a monophonic, single loudspeaker basis (preferably mounted with the same spatial relationship angle to the listener as the final use model) or dual loudspeakers with mono program, to evaluate tonal aberrations independent of spatial effects, but the final evaluation must be done in the actual, use model, dual loudspeaker mode.

Generally I recommend starting from a flat axial and well behaved off-axis response as a first step, to isolate any system resonances, driver interactive effects, diffraction issues, or other a problems before optimizing the system power response. There are many exceptions, wherein the transducers are purposely designed to have a built in ‘effect curve’.

The goal of an compensations I am discussing here are an attempt to address universal objective issues, not for the purpose of subjective personal taste enhancements.

If one were to consider adjust a loudspeaker to personal taste, I would recommend getting everything objectively correct first, so that any preference distortions that one adds are easily adjusted and identifiable, instead of just being a pleasant artifact/distortion of the loudspeaker that is left in tact. This way, one can control the system more easily, having full understanding of what is being added, with the ability to get back to an objective baseline.

If you are seeing this dip single ground plane, outdoor measurements, it may be misleading. A single boundary reflection is often difficult to eliminate completely, but the room has many boundaries, and if the loudspeaker is purposefully placed for an optimal ratio front wall, floor, and sidewall distances (and ideally, ceiling distance also), then the boundary reflections will average out in a manner that will substantially eliminate any severe dips as you observed in your outdoor measurement.

Applying absorbent to the front and sidewalls will help further, as will some form of absorbent or diffusor on the floor. All of these tactics used together will tend to smooth the ripple to an acceptable level.

Only having a single reflection boundary is rather artificial. In fact, in some room situations, if you were to eliminate all the reflections 100%, except having one strong reflection remaining, it would sound worse than if you had all the reflections distributed in an effective manner, such as establishing golden ratio (~1.6) secondary path-length ratios.

Correlated reflections are psycho-acoustically the most disturbing (those arriving equally to both ears, such as floor reflection, front wall reflection, and ceiling reflections) particularly if just one dominates.

Make sense?

- James

Yes it makes sense. One can really understand how the "established" HiFi-brands arent into these kind of compensations.
But I find it to be a pity aswell. Feels I bit of a shame that companies like B&W or Revel spend all that money on designing new high-tech transducers and put them in advanced boxes with "high-end" expensive crossover parts only to have a sub-optimal crossover that doesnt make the best of the stereoflaw.
On the other hand those companies seem to have the best ability to incorporate the right compensations. Would you say the concept of the stereoflaw is to difficult a matter for "normal" people(even HiFi-nuts)? As always I assume its the salespeople that dictates what can be done.

Regarding the compensation curve I posted above Im avare that it isnt as simple as that. And I apologize if thats how it seemed. I just brought it in as an example of how it could look like. Being a generalist I tend to speak in general terms even when I know it isnt as easy as that. :)

I guess these compensations are percieved differently if its a big speaker or a small. Different placements of the speaker in the room would also affect this. So a firm advice on placement should be a minimal when it comes to loudspeakers, wonder why the industry are so afraid of that..

What are your feelings when it comes to multichannel sound? Are these compensations still good to have or should they look different (not exist even)?
What are your views on multi channel sound in general. Reading Tooles book I got the expression that he seem to think multi channel are the way to go forward? Seeing how difficult it is to make a stereo pair work in an optimal way, it feels daunting to have 5 or more speakers with the "right" compensations applied. Or maybe it isnt all that critical?
The lack of an stardard when mixing surround sound it might be hard to generalize?

Kraniet,

I understood that the curve that you showed was merely to provide a generalized example to give context for discussion. I did not see it as you reducing it all to just frequency response.

I am sorry if you felt criticized for bringing forth a good question, which you did in a clear and useful manner. You should always feel free to come forth with questions that are meaningful to yourself.

••••••••••••

As you suggest, sales and marketing often tend to drive the product decisions more than the engineers. The general aspects of the stereo flaw are not difficult to convey or understand, but to comprehend all the complex interactive details and what to do about them is a much more complex issue.

Two of the main impediments are; 1) there is a wide range of recording processes, making it difficult to coordinate with all of them, and 2) the desire to not make loudspeaker use models too specific, so that they can fit into more homes without impacting the sensitivities to interior design preference.

In terms of #1, relative to an ideal angular loudspeaker alignment, I would have to say that, unfortunately, there is not just one ideal arrangement. Numbers like 21 degrees or 23 degrees have limited meaning without carefully defining all the variables.

Due to the variation in encoding process, the recording may be encoded by the pure definition of binaural, or by any one of the almost unlimited permutations of what is called the “stereo” recording process. Again, it is only one part of the picture, but one variable that is not static, but rather dynamic, depending on the recording/playback processing, is that of optimal listening angle, which is reduced to a very low number for binaural recordings played through a crosstalk cancellation system and intermediate number for “stereo” recordings with an imbedded level of crosstalk cancellation, to a bit wider for standard stereo, still wider for certain mixes that include center channel. The particular angle numbers can be defined more precisely as one more clearly defines the myriad of associated parameters.

••••••••••••

I always find it remarkable that while a first rule for marketing is to have a product that is distinctive, with advantages over the competition, but at the same time most marketing departments will balk at anything that appears too different or unusual. They want to be differentiated and better, while at the same time remaining safely similar to the competition.
Strange.

•••••••••••••••

Your last sentence speaks to the truth of one of the most significant problems with all forms of recording/playback process.

I’ll take a risk and state that my current feeling is that multi-channel sound (meaning more than two-channels) as it stands today, and the way it is most often applied, has little value for anything other than home-theater applications, unless one is just looking for merely adding artificial sensationalism to a music program.

With home theater recordings there are at least dubbing stage standards that allow one to substantially replicate what the recording engineer is hearing at their seat in the mixing stage and for that mix also to have some correspondence between one’s experience at the movie theater and in one’s own home theater.

But for music, it is currently a rather confused state of affairs.

Mr. Toole may be correct in stating that it is the way forward, but at the very least, we must move very far forward from where we are today, before it reaches the level of becoming “the way”.

There is good work being done in this regard, but it is still in its formative stages, technically, and even more so, politically.

That said, I remain optimistic about what we can achieve.

All the best,

- James

[IngOehman]

Regarding the compensation curve I posted above Im avare that it isnt as simple as that. And I apologize if thats how it seemed.

No problems from my horizon, as long as I, mentioned by name as I
was, is allowed to clarify that I do not see it that way, nor have ever to
my kowleadge, presented it as merly a timbral property.

Actually, it is a good thing that you bought it up, since one persons view
of a matter, is often shared by others - and this way I was given a chance
to clarify that I have never spoken about phenomenons in one dimension
only.

Ana again, I'm happy as long as I get the chance to clarify that I do not
want to be hold responsible for something I have neither thought nor
said. So thank you for that chance!

I just brought it in as an example of how it could look like. Being a generalist I tend to speak in general terms even when I know it isnt as easy as that. :)

That is fine by me, as long as I'm not hold responsible.

And again I like to remind everyone that the compensation does not only
have a lot of different reasons and is therefore different in different speak-
ers - it is not a mere timbral question either.

I guess these compensations are percieved differently if its a big speaker or a small. Different placements of the speaker in the room would also affect this. So a firm advice on placement should be a minimal when it comes to loudspeakers, wonder why the industry are so afraid of that..

I believe it is a combination of three things:

1. These are relativly small errors compared with most of the actual prob-
lems that has to be dealt with just to arrive at an uncompensated speaker
of descent quality.
So if I had to suggest an order of priority, the compensation of the stereo
system flaws would come in, long after matters like controlling distortion,
dispersion, dynamik linearity, and having the speaker function in a room.
And already these are more than a handful to deal with. So in my opinion,
very few loudspeakers are suffering most from "not having stereo system
compensations".

2. The main part of the speaker industry has not even reflected over mat-
ters of stereo system flaws at all, and the very small part that has - are
either unclear over their own views on it (due to an open mind in combo
with their lack of studies on it), or they do not agree with others.
So there is really no general answer available, that can be applied for any-
one who "is not afraid" to apply it.
Finding a sollution fitting ones philosophy in all other aspects of reproducing
sound timbrally and in space and time, can take decades of research.

3. Some loudspeaker manufacturers are, due to their specific philosophical
ideas on how to communicate the recorded information to the listeners,
not really in NEED of applying any compenatios at all - since they do not
manufacture loudspeakers that works in a manner that will make the flaws
appear anyway, so there is really no need for compensations!
My view on that is, good for them!

What are your feelings when it comes to multichannel sound? Are these compensations still good to have or should they look different (not exist even)?

What are your views on multi channel sound in general. Reading Tooles book I got the expression that he seem to think multi channel are the way to go forward? Seeing how difficult it is to make a stereo pair work in an optimal way, it feels daunting to have 5 or more speakers with the "right" compensations applied. Or maybe it isnt all that critical?

The lack of an stardard when mixing surround sound it might be hard to generalize?

I understand that the above questions are not directed to me, so I say
no more.


Vh, iö

[IngOehman]

Hi everybody!

Just for the protocol, I'd like to clarify that all the problems which I include
in what I like to call "the stereo system flaws" or "the intrinsic flaws of the
stereo system", are problems in MANY different domains.

I.e. both the timbral domain, imaging distortions, dynamic (and pseudo
dynamic (linear behaviours that still reduce dynamic range)) effects and
also time resolution loss effects.

I say this, since I sense in the above quoted, that Kraniet has reduced it
all, or close to everything to being a question of frequency response - and
even to the idea that a single univocal target curve could be "the answer to
what constitutes as a correct compensation".

Nothing can be further from the truth.

All these things are delicately entangled in quite an illusive and convoluted
manner...


I do not like simplifications, and I really do not like to have things I've said
simplified by others - and then still being hold responsible!

I'm not responsible.

(Actually - I'm one of the most irresponsible normalized earthlings that I
know. I am to be trusted about everything - but with nothing.)

- - -

The problems also occur both during "encoding" and "decoding" (recording
and replaying the recording), both are exhibiting differens set of problems,
often cross-depending in intricate patterns, which actually creates quite a
few interesting possibilities to optimize all balances.

Anyway; the possibilities to find and optimize "the stereo system compen-
sations" are no less multifaceted than the originating problem is. And thus
the compensations (optimally applied) are also addressing the behaviours
in ALL of the mentioned domains.

- - -

For practical reasons, I try to avoid going into detailed descriptions of the
problems and solutions on internet foras.

BUT - I believe that I have been very clear (also here on faktiskt.se) that
even if you ONLY look on the timbral problems (ignore ALL the problems
that are manifesting in the other domains) caused by the stereo system
(i.e. by trying to pack the complexity of real life multi dimensional sound
into only two one-dimensional channels) and also ignoring* all radiating
directions other than the one aiming directly at the listener - I still do divide
both the problem and the solution in 12 different components - resulting
in quite different 0 degree frequency response curves, depending on all
other parameters of the loudspeaker.

So there are no single target curve, and I do not see how there can even
be one! The idea of a single univocal target curve goes against everything
I know about the inner workings of our hearing.

What can be, and I believe is - is a complex equation that can be used to
create a target curve for one specific loudspeaker (assuming the environ-
ment is reasonable predictable).

I use such an equation, but again - there are at least 12 different aspects
of it where 11 are variables! (I include the 12 important enough to play a
role that is larger than the uncertainties.)

And again - the timbrel part of the stereo system flaws are only a minor
part of the everything that is caused by the stereo system.

- - -

I just wanted so say that, so that no one attributes the idea to me, of the
stereo system flaws possible corrections being nothing but a target curve
in the timbral domain.

This said, I hope that no one misunderstands what I just wrote to be an
attach aimed against the BBC-dip, or the men behind it. To the contrary,
to my knowledge, the BBC's studies ware amongst the first in the word
addressing problems of the stereo system itself, and though being just a
fraction of a fraction of the truth, it does not diminish the fact that a first
step is often the most important - since it leads the way by pointing out
the direction.


Best regards, Ingvar

- - - - -

PS. Please excuse my English. Writing in English make me feel like my
head is full of brake fluid. Well, not that I've tried that...


*When I spoke about ignoring other radiation directions, I did not really
mean ignoring it (it is a vital part of the equation) only ignoring having it
adjusted separately - as a simplification, to point out that it is still very
complicated, even after such a simplification.

In real life however, engineering loudspeakers, I do no such simplifications,
but try to control everything and give each parameter the properties I like
it to have to work well in the application.

Not just "see what I got" and then try to do the best of it in regards of the
stereo system corrections.

Hello Ingvar,

I both agree and empathize with you regarding the concern and danger of having ones forum comments reduced to oversimplifications.

The multi-dimensional aspects of effective endeavors in loudspeaker engineering are most often far more complex that what can be reduced to a few paragraphs, but, we are a social animals with a shared passion, so we embark on a dialog about the things we care about.

So true.

It seems that maybe the best one can do in this type of forum is to entertain, bring some joy and to provide incomplete ideas that hopefully at least point in a direction that empowers others to be a little more effective at discovering their own answers….and in the mean time, hope that one’s statements aren’t terribly misconstrued as they are passed on and repeated.

Your words are wise, but I do not like to rely on hope.

So if possible, I always try to communicate not only the answer to an
isolated small question (in a large complex), but also the "complexity of
the whole" - if so only by reminding everybody of it, with a few words.

It is not always enough.

But then again, there is often neither time nor room for more.

Anyhow, for my part, I appreciate your use of English and tolerating the sensation of a brake fluid filled head.

Well what can I say - you can try it by switching to Swedish.

All the best,

- James

Same to you, James.


Vh, iö

[paa]

James, when you try to get a rectangular box like The Clue as coupled as possible to the wall, how much absorption would you prefer behind it? What size and thickness would you consider minimum and optimum?

As a minimum I recommend 75mm thick acoustic foam that is arranged on the wall, inward from the right loudspeaker (going towards the left loudspeaker), starting at the inside corner of the cabinet that is closest to the wall, and also starting at the bottom edge of the cabinet.

The panel should be about 60 x 60cm, extending about 25cm above the top of the cabinet.

Ideally, additional amounts and placement would be adapted to the needs of a particular environment.


For a more complete/optimal arrangement, as a general rule, I would suggest a 75mm thick, 150cm x 150cm square piece centered horizontally behind the cabinet, and starting at the floor.

Also, place 100mm thick acoustic foam outward from the speaker, (to the right of the right loudspeaker) over to the corner, and from the corner extend 1/3 the length of the sidewall, OR, a piece 1-meter wide centered on the point where the first specular reflection bounces off the sidewall to the listener.

Again, each room has it’s own issues to deal with, but these are the general guidelines for minimum and optimal absorption.

Let me know if I need to provide further explanation.

Cheers,

- James

I may be thinking slowly, but now I wonder if the more full version of room damping you suggested here will make the loudspeaker more forward radiating, almost like cardioid speakers
Maybe the bass loss from cardioid boxes can thus be considered unnessesary?
Would you like to comment on this?

[vr6jos]

Nu finns ett par till salu. Någon rik herre kan väl köpa bara för skojs skull och recensera!?!

http://www.hifimagasinet.com/annonsDetalj.asp?annonsID=53611