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Distortion down to -300 dB, what exactly does that mean physically?

peterzuid

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What are you even talking about? I've designed digital filters since the late 1970's. I've published papers on the subject, as well as papers on the audibility of artifacts due partially to filtering, etc.

I can't even tell if you mean "complex filters" (meaning coefficients with an imaginary part), or "complex filters" as filters neither minimum phase or constant delay, or what you mean, and I'm inclined to think you don't know, either.

What does "high tap filters" even mean? I asked you before, and you chose to engage in professional disparagement rather than answer the question.

SO WHAT DO YOU MEAN BY "HIGH TAP FILTERS". Allow me to make one thing clear, IT'S NOT A TERM IN THE ART.

After you cope with that, we can figure out what you think you actually meant.

So, what are "high tap filters"? Speak up, answer the question, and stop insulting others.

As to 300dB dynamic range, yes, that's ridiculous. The SPL of 1 atmosphere peak signal is 194dB. The noise floor of the atmosphere in the audio bandwidth at the eardrun is 6dB SPL or a bit more.

Learn for once.
Hi j.j, I don't think I used this term "high tap filters" anywhere in this forum. Rob Watts mentions tap length and I can give you my understanding of it. The perfect analogue reproduction from a digital to analogue conversion will be the one with the sinc function as an impulse respons. This is impossible to design as this is a time unlimited respons and the exact analogue output between two samples has to be calculated with infinite precision and timing . The M-scaler uses "only" a time interval of ca. 1 second in which the contribution of all samples in that second is calculated to define the output level at a certain moment. In addition, the M-scaler defines 14 extra output levels (in case of red book CD) between two input samples. It does so by calculating the contribution of all digital input signals in a frame of a second. The contribution of all samples will be calculated with sinc function truncated or a somewhat "Watts modified sinc function" That's why longer tap lengths are necessary to be used.
 

AdamG

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Thread notice: Warning(s) have been issued for personal insults and excessive rudeness. Any further violations of this nature will receive an Official Warning and permanent Thread Ban. Please and thank you.
 

Geert

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The perfect analogue reproduction from a digital to analogue conversion will be the one with the sinc function as an impulse respons. This is impossible to design as this is a time unlimited respons and the exact analogue output between two samples has to be calculated with infinite precision and timing .
Why do we need "infinite precision and timing"?
How would the sinc function not being "infinite precise and timed" manifest itself in measurable parameters?
Are there no hearing thersholds that apply to these parameters?
 

egellings

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300 dB--an impossibly large ratio for any two things in real life.
 

phoenixdogfan

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Imagine a gnat's wings beating from a thousand feet away, now imagine a supernova exploding this very second right next to your ear...
 

DonH56

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amirm

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I would like to get your D/A conversion system and put this on a test with the Chord I own. Gladly to pay whatever in case it sounds better.
Do you know how to determine if it sounds better?
 

amirm

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The M-scaler uses "only" a time interval of ca. 1 second in which the contribution of all samples in that second is calculated to define the output level at a certain moment.
JJ is one of our luminaires with signal processing one of his core specialties. So please don't try to explain things to him this way. It is liable to get your nose bloodied. What you just wrote is just the basics of how such a filter works which by the way, results in high latencies. Best to not watch any videos with it unless you can delay the video by the same amount.
 

audio2design

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The perfect analogue reproduction from a digital to analogue conversion will be the one with the sinc function as an impulse respons.

Perfect is the enemy of good. It is also the enemy of reality.

The audio world is all "a buzz" with talk to FPGA's like they are some new fangle dangled technology. I was putting digital filters and processing in FPGAs in 1989 for video and audio. FPGA is a buzz word in audio to impress those that don't know any better. No audio is not unique to marketing speak, but it does seem to have a larger (purchasing power)/(knowledge quotient) fueled primarily by ego, both for the need for special skills, and the need to be "special", not unlike other luxury goods. The difference in audio is it is not the luxury that is marketed, but perceived real improvements.

You will note the lack of very easily verified measurements from the supplier for these high end and expensive scalers. Even when done by 3rd parties (going off memory), the difference was almost nil, and that difference would be highly DAC dependent and given questionable implementations of some esoteric DACs, there is no guarantee that feeding them a high sample rate signal, even if mathematically slightly better than what they would do internally, is going to be better and not worse. Sure, if you have convinced yourself that a non-oversampling DAC is a great idea, then any scaler would be an improvement. Sometimes painting a pig is a good idea. I could have achieved the same thing for a few dollars of hardware in the DAC, and put in the common sense over sampling, even if just a user option.

Also note the lack of a clear mathematical analysis providing, using a typical DAC chain, what the likely real benefit is going to be. Again, something pretty easy to do.

Your statements above will "fly" on many audiophile forums where the average, heck the vast majority of users both don't have a grasp of the technology, but worse, really don't want to understand it unless it supports their desired perception of how things work. Is that harsh? Perhaps, but I have taken enough abuse, as humorous as it normally is, on those forums, to know this factually. I could find a 100 posts just today on likely one forum of members revelling in their ignorance and other members lapping it up because it fits what they want to believe. I kid you not, I have had an audio charlatan supplier offer money to Dox me. You would think it would have been easier to address my arguments if one could.

Yes, there are people here who will confuse "accurate" with "must sound good", and there are dogmatic views about how things must be, and yes, my personal favorite, the dogmatic view that "accurate signals = accurate sound", which totally ignores what happens on the way to the brain making an interpretation. HOWEVER <-- Highly emphasized <-- There is a big difference between "sounds better" whatever that means, and "sounds different", or should I say "has any chance of sounding different". The delusional audiophile mantra, "everything matters" is simply poppycock. Everything does not matter. A whole lot of stuff does not matter, and that is where we are with your posts.
 

egellings

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As for the sound of the gnat's wings at that distance, there would be no sound waves at my ears; random air movement / noise would erase that. As for the supernova, I would not hear that either, since sound does not travel in a vacuum. So comparing their sound intensities as heard by a listener somewhere on earth is meaningless.
 

j_j

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... Rob Watts mentions tap length and I can give you my understanding of it. The perfect analogue reproduction from a digital to analogue conversion will be the one with the sinc function as an impulse respons. ...

This of course a ridiculous choice of course for doing reconstruction. One can design either a constant delay, or mixed delay filter of quite finite length for any purpose, while providing reconstruction to any arbitrary level, say, under 1 LSB, without having to resort to sinc functions, etc. Look up "remez exchange" for one well-known way to design a finite length filter that will give, assuming you feed it properly (never assume people know what they are doing) the best filter one can achieve for a given length.

Then you look at the results, and either make the filter longer (if it's not good enough) or shorter (if it's excessively good) or you simply allow yourself to be satisfied with over kill (the usual way to do this).

It's really not hard. If you must do some kind of resampling (say by 14 as you mentioned, although 14 is pretty silly, it would be much more efficient to do 16, Crochiere and Rabiner showed how to do this to any arbitrary cleanliness, efficiently, back in the very early 1980's. I know, I worked for them, some of my work is in the book.

https://www.amazon.com/Multirate-Digital-Signal-Processing-Crochiere/dp/0136051626

If you want to do variable-rate resampling, Proakis shows how to do that to any arbitrary level in one of his early books on digital signal processing.

These are almost hoary-old methods, and they work JUST FINE. There's no need for this "sinc" stuff, you're spouting theory that is both inefficient and unnecessary there. (Yes, the theory is right, but so is the theory of filters and wavelets.)

Why do we need "infinite precision and timing"?
How would the sinc function not being "infinite precise and timed" manifest itself in measurable parameters?
Are there no hearing thersholds that apply to these parameters?

Well, precisely. Given the basic atmospheric noise that nobody seems to want to remember, there's no such thing as "infinite precision". That's really a very basic point here, basic atmospheric noise due to the particulate nature (i.e. molecules, argon atoms) makes any kind of "infinite precision" nonexistent, timewise, levelwise, you name it.

(Not directed at Geert here)

Furthermore, air propagation at high levels (above 120dB or so, detectable, barely, maybe, at 90dB, by an accurate machine) is NOT LINEAR. So enough with these silly "infinite precision" arguments. If you want to argue about multistage interpolation and decimation, your first job is to prove that Gauss was wrong, and the Fourier Series does not converge in L2 with finite power signals. Good luck with that.
 

audio2design

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Well, I suppose there would be "sound" for a bit while you are vaporized by the radiation. :D

Doubt you'd hear it, or know.


I always wondered where "people" go when vaporized by a phaser in Star Trek. Can you imagine the explosion from turning 200lbs of human into vapor in < 1 second? :)
 

j_j

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I always wondered where "people" go when vaporized by a phaser in Star Trek. Can you imagine the explosion from turning 200lbs of human into vapor in < 1 second? :)

Yeah, I've asked that question, too. The body being mostly water, every 16CCs of a body will turn into 1.5*22.4 liters of gas, using the "all water" assumption. And then you are left with the problem of an exactly stoichometric mixture of H2 and O2 at a very high temperature in a very small space. Something's gonna happen!
 

RayDunzl

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We first need a volunteer for the experiment to have any hope of success.
 
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