Is *signal purity* BS?

Hi Joel

I didn't exactly try to convince you Joel. Why should I?

I just put an opinion here. :)

Yes the Brugmann-man is clever, but reading his white paper it is to *cheap* to convince people about his theories on passive x-overs when using Butherworth transfigurations as an example.

It should be known to speaker guys that Butherworth is worth nothing. NOTHING!

What do you think would have been his conclusions if he compared his theories to a 1:1 transfiguration?

So this is also BS *theory purity*

Gerner:)

The same would apply to any min phase crossover transfer function you care to mention. Doesn't matter whether its Linkwitz, Butterworth or whatever else.

Its really a short paper on IIR vs. FIR filtering with a few of his correction algorithms thrown in.

EDIT: There's also an error in that paper because the butterworth filters aren't showing the correct transfer function that a genuine butterworth network produces.
 
Hahaha shin...so you got Aleksandars cookbook for controling magnetism and stray flux...

He is a lovely guy and I typically call him Nikola Tesla the 2nd. You know why if you met him.

I will bring him your greetings, well if it's ok?

Yes, give Alex my regards when you see him.

There's many bright folks in audio but Alex is one of the brighter ones and also takes time out to take an interest in small independent efforts such as that of a DIY'er.

He's got himself a steady customer here and I'll certainly be using his HF driver expertise for my next project in '08.

I appoligize for not commenting your beautiful loudspeakers. They are damn beautiful but not only that. I can see you have some pedigree in speaker building as well.

Thankyou and likewise on your own design.
 
Its a little misleading but a 64bit operating system isn't required for 64bit audio processing. You don't even need a 64bit processor although one with 64bit extensions will be roughly twice as efficient should the code take of advantage of that.

64bit data can be represented by two 32bit types and really when you see 64bit marketed on CPU's and OS's they're talking about the memory and data address range which is a physical quantity. You can represent 512bit data and process it provided you have the software coded accordingly. Of course you take a performance hit over processing that same data using a bonafide 512bit instruction set because, say a 32bit CPU, has 16 times less physical address bits so the data needs to be split and sent in 16 chunks compared to just once on genuine 512bit CPU. Very inefficient.

Just for the record I use Vista 64bit with an Intel quad core running at 3Ghz.



If digital then 16-24bits input, this is then processed with 64bits of resolution at either native sample rate or can be upsampled upto 192Khz. If analogue input then its converted to digital using the 24bit ADCs and then processed.
Once any processing is finished its back to the DAC which are 16/24bit so dithering occurs. So in a nutshell 64bit precision is used throughout the processing to maintain sufficient headroom for iterative computational stacking and minimise rounding or truncation. The end result is more transparent processing with less digital footprint.



The software is written by Dr. Ulrich Bruggemann. A very bright chap that has worked with room correction for some years including solutions for Tact.

Hi Shin

First thanks for briefing me.

I know from my PC-player world that i can double the data as *many* times as I would care for.

You gave me the answers. I do not need to jump for a 64 bit MOBO where no programs available hardly can work. But it will be useful for the course of processor duties. Right?

I have a 32 bit MOBO with quadro processor. + a bunch of PC's and Laptops. Yes 6 of them.

Yes I know Dr. Ulrich Bruggemann has been contributing to TACT.

I just pointed out he could have illustrated his enourmous capabilities and knowledge, not holding up his theories against a stupid Butterworth equation. He came around that part a little too easy.

He could at least compare his program to this:

http://www.linkwitzlab.com/crossovers.htm

and we all know this as well:

http://www.linkwitzlab.com/stereo reproduction.htm

this is another approach to filtering not hurting my BS, but still adds components in the chain and remove fewer in the passive filter:

http://www.linkwitzlab.com/filters.htm



And no, I do not link up to Siegfried. Don't know the guy at all.


Having said that, I have the deepest respect for Dr. Ulrich Bruggemann.

I will give his program a try and report back when I am done.
That may take some time..:D

Thanks once more Shin and Mike.

Gerner:)
 
Let me know when you have a chance and you are welcome here of course.

Thanks endust....are you the guy in Budapest...and then know Ferenc?

I just ask, because all those nick-names on forums are hard to identify who's behind.

Gerner:)
 
The same would apply to any min phase crossover transfer function you care to mention. Doesn't matter whether its Linkwitz, Butterworth or whatever else.

Its really a short paper on IIR vs. FIR filtering with a few of his correction algorithms thrown in.

EDIT: There's also an error in that paper because the butterworth filters aren't showing the correct transfer function that a genuine butterworth network produces.

I noticed...

Gerner :)
 
Yes, give Alex my regards when you see him.

There's many bright folks in audio but Alex is one of the brighter ones and also takes time out to take an interest in small independent efforts such as that of a DIY'er.

He's got himself a steady customer here and I'll certainly be using his HF driver expertise for my next project in '08.



Thankyou and likewise on your own design.

I shall hug Aleks from you.

He is my cup of tea. :)

What we are to try out is not exactly a DIY project...but beyon that scale (positively meant of course).

But if it works out well, we are talking about a completely new ribbon which combined represent a 45 x 160 cm 216 ribbon 16 x 200 mm horisontal bended ribbon surface element, enclosing 20 kg's of neodymium and a new magnet structure without visible anchor and a new suspension technique. Flux control above anything seen before. > 1 Tesla in the gab.
Efficiency around 107 db/w/m. Cut off at 500 Hz. No trafoes involved.
On top of that it will be an open baffle system.


Oh... the basses...wait and see. Hehe :D

Gerner:)

Edit: The Swings is not my design, but my theories are found inside there. The design and all rights belongs to Bert.
 
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I shall hug Aleks from you.

He is my cup of tea. :)

What we are to try out is not exactly a DIY project...but beyon that scale (positively meant of course).

But if it works out well, we are talking about a completely new ribbon which combined represent a 45 x 160 cm 216 ribbon 16 x 2000 mm horisontal bended ribbon surface element, enclosing 20 kg's of neodymium and a new magnet structure without visible anchor and a new suspension technique. Flux control above anything seen before. > 1 Tesla in the gab.
Efficiency around 107 db/w/m. Cut off at 500 Hz. No trafoes involved.
On top of that it will be an open baffle system.


Oh... the basses...wait and see. Hehe :D

Gerner:)

Edit: The Swings is not my design, but my theories are found inside there. The design and all rights belongs to Bert.

In Jan 07 Alex sent pictures of a prototype he was working on. It consisted of multiple dipole ribbon elements in close formation along with neodymium slugs built into the framework/magnetic circuit. The whole thing was on a pivot mechanism that could be adjusted to create a wider dispersion pattern.

I don't think he ever finished it as he got busy with more important commercial developments. It certainly looked interesting though.

I was thinking about having something custom built from him for my next attempt. Romy seems pleased enough with his teardrop 109dB high efficiency ribbon although that's not the direction I'd be interested in going but does show Alex' designs are fairly flexible.
 
In Jan 07 Alex sent pictures of a prototype he was working on. It consisted of multiple dipole ribbon elements in close formation along with neodymium slugs built into the framework/magnetic circuit. The whole thing was on a pivot mechanism that could be adjusted to create a wider dispersion pattern.

I don't think he ever finished it as he got busy with more important commercial developments. It certainly looked interesting though.

I was thinking about having something custom built from him for my next attempt. Romy seems pleased enough with his teardrop 109dB high efficiency ribbon although that's not the direction I'd be interested in going but does show Alex' designs are fairly flexible.

Shin, yes this is the THING, but further developments are applied since those pics were published by Aleks.

The efficiency is still to be determined soon. It might as well be 112/db/w/m, as we don't know yet what the stacking will do to us and when the imp. ratio of the trafo is out of the game, only maths can tell. I was very concervative naming the efficiency. Undersell----overdeliver is my codex.
But let's see.

What is important is, that this huge surface flash fast multiple element can go so low before high-pass filtering is applied. Amazing...:)

A two way system showing piston qualities and giving a damn about time alignment. At least for the ribbons part.
Basses will be averaged.
I think *you* know what I mean.

But if it works out well, then it will be a comercialised system and not a DIY.
Pricetag > at least 100.000 £ :MILD:

Cheers....

Isn't it about time to celebrating a little now...???

You have very little left of 2007 your place.

Gerner :)
 
To put it shortly, the main purpose of the "shortest track" or "pure signal" is preserving the microdynamics which are the weakest link of most good systems. Screwed microdynamics, above all, rob the music of it's emotional context. EQ deals with other things, mainly freq. responce. But the thing is - we humans adapt to tonal deviations much more easily than to dynamic ones. That's why equalised music often sound "right" but dull. I have used it all - from parametric to TacT, and have no intention to come back to it.
 
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Why would you want something other than the purest reproduction of what's in a recording?:rds2:

• Because many recordings are poor, overly compressed, and need some kind of "tweaking".

• Can that be done without adding harmfull distortion? (I don't object to a little "euphonic" distortion as long as it makes the poor recording better)

• Well, I can't answer that :D... But you can either use some kind of tone adjustment or a "flavoured" piece of gear.

• This second option seems to be the worst alternative since it might have a negative effect on good recordings.

And how about "contrast"?:confused:

• Could you add contrast to flat and liveless recordings by replacing, say, the speakers?

• Or maybe have two systems with different flavours?
___

Of course all this BS only applies to those who wish to listen to the music...

Those who would rather listen to the recording should stick to ruler flat response gear.:Quad:

P.S.: My gear goes for the smallest number of parts, shortest signal path to, as Nomoretweaks says, "preserve the microdynamics"
 
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To put it shortly, the main purpose of the "shortest track" or "pure signal" is preserving the microdynamics which are the weakest link of most good systems. Screwed microdynamics, above all, rob the music of it's emotional context. EQ deals with other things, mainly freq. responce. But the thing is - we humans adapt to tonal deviations much more easily than to dynamic ones. That's why equalised music often sound "right" but dull. I have used it all - from parametric to TacT, and have no intention to come back to it.

excellent thoughts, pretty much mirroring my recent findings. superior microdynamics are the main reason IMHO why some unlinear speakers (different fullrangers in the first place, and some horns to lesser extent) sound so exciting and emotionally superior despite a flawed frequency response.

i've recently had a thrilling demo at teresonic, a manufacturer of rather excellent lowther based cabinets - in their system (consisting of their own CD player and 300b IST SET with a mercury vapor rectification), the signal is passing only through 7 elements in total (valves, resistors, transformers, not a single cap) between the laser pickup and the membrane of the lowthers. a truly remarkable sound full of explosive microdynamics and emotional content: you just stop thinking about lowther shout, lack of deepest bass etc.

nomoretweaks, would you mind sharing us details on your system (in particular - loudspeakers)
 
Those who would rather listen to the recording should stick to ruler flat response gear.:Quad:

P.S.: My gear goes for the smallest number of parts, shortest signal path to, as Nomoretweaks says, "preserve the microdynamics"


And that is a perfectly valid method. Good on you for being so passionate about it.

Its all hot air talking about what works and what doesn't and it also seems a number of folks here haven't heard EQ done well. Rather than continue down a well beaten path, I'd be happy to have any fellow enthusiasts as guests should you wish to hear a decent implementation of digital room correction. The only real way to judge a method is to hear it done to a high standard.

An observation; I remember about a decade ago when audiophiles were strongly adverse to physical room treatments. They are now seeing their way into the best setups around the world. Maybe in another 10 years time the same will be true of digital crossovers and correction. Already studios are using this technology just as they adopted physical treatments early on.

BTW Happy new year to all :)
 
" Is Signal Purity BS ? "


Pretty much everything written by self aggrandizing audiophiles is BS.


It's a wallet orientated cock fest of bollocks.

Me included ;)
 
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