John Curl's Blowtorch preamplifier part II

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Thanks for that link, Scott.

... but ... log3(60dB)? (from page 7)

To borrow an acronym from Thorsten: WTF?

We regularly see log10() for dB ratios, log2() for efficiency with binary processors, and the natural ln(), which is log2.718(), but I've never seen anyone attempt to make practical use of log3() before this paper. 20*log10(2^10) == 60 dB, showing that you need 60 dB attenuation for the AAF before a 10-bit A/D, but how does that lead to log3(60dB)?

Also, I calculate log3(60) as 3.73, not 6.28

What am I missing here?

Don't know, why do physicists and EE's use different scaling on FFT's? I have not read it carefully enough to see.
 
Paul Frindle, Are we Measuring the right Things? Artefact audibility versus measurement
The Measure of Audio, AES UK Conference 1997.

I have to dig deeper in my archive (which suffers in the moment from similar problems as jneutrons), but afair Theile and Wittek should have done some experiments with ITDs in stereophonic presentations, which confirm at least the ~10µs .

As stated earlier, small head movements actually increase the localization accuracy, investigated a couple of times with good accordance.

BTW, this was an interesting description in last years discussion:

Gearslutz.com - View Single Post - Paul Frindle - Is This Truth Or Myth? -

later on Frindle mentioned differences detectable as low as 4µs.


I'm glad you mentioned Paul Frindle because it was my intention to mention his work on the SONY OXF-R3 console (which is considered a state of the art digital console costing over 1 million$).

He has a deep understanding of analog and digital and he did some serious research to correlate audibility of various processes.
He discovered that even dither at -110dBFS could be detected in -70dBFS tape noise recorded digitally if in the randomness of the dither there was a partial statistical correlation. Let's not forget that there is no real random in digital computer world, so you have to know how to design a good random generator using limited computing resources.

chrissugar
 
Hi,

Sorry I'm done with this. As John points out the concept of subtractive dither is over 50yr. old in image coding etc.

That is completely besides the point.

So you insist on not showing anything real that relates to how things really work while presenting something that is an idealised version of hos things could be done, but are not. That is your prerogative...

Ciao T
 
Hi,

All the discussion about dither being evil is absurd.

Is it? Would you say that applying 14 Bits of worth of dither and converting a 24 Bit PCM Signal to 10 Bit with this is a good idea?

Dither is absolutely necessary to avoid distortion that would appear if high bit depth files would be simply truncated to 16bit.

Just because you say so does not make it so.

You are aware of Ethan Wienders tests?

Grekim and Ethan test dither, jitter, A/D

It is very simple to experiment by generating a 24bit sine wave in some proaudio software (Wavelab) and truncate it to 16bits and also dither it to 16bits. Just do an analysis and also listen. You can draw your conclusions.

For music the truncated version will sound distorted and all the low level information like space cues and reverb tails will be lost while the dithered version will retain all the low level information and no distortion.

Funnily enough, that is not what I hear and what for example Gremkin/Wiener found.

And note, I do take umbrage as such with 2LSB dither applied to the 16 Bit CD Release when coming from a 24 Bit Master (though the actual need for said is debatable), but I do take issue with 16384 LSB Dither applied to a 24 Bit signal...

Ciao T
 
Hi,

BTW, how about you show instead the simulation of a real system, instead of making up stuff that we still have not even a feasibility analysis for. I mean someone may mistake what you posted for an actual illustration of what actual dither does in an actual 16 Bit System...

It is actually already quite obvious in your pictures...


I missed the return to the condecending tone. What I presented was text book stuff. I would think converters used to master CD's are a bit better than off the shelf and their INL and DNL are pretty close to ideal, in any case things are much better now. If you sent a complete map of the INL and DNL of your particular A/D or D/A it could be simulated, but no I don't have time.

As for the harping on FFT's, ideal TPDF dither produces uniformly distributed white noise as a residual, the spectra are not interesting on any scale. Text book stuff again.

No need to use real DAC's, all this can be done using "idealised" converters and in something like Mathlab. I lack the time to set this up.

Why am I not surprised? It's either that or someone else paid for it so it's proprietary. Gee when I use "idealized" converters it's not OK.
 

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as cited before, Paul Frindle reported in his Convention paper ABX-Results for an impressive list of small differences (done in studios) including a constant halfsample width delay of one channel (leads to ~11µs) and confirmed it later several times in discussions over at the pgm list.

Paul Frindle, Are we Measuring the right Things? Artefact audibility versus measurement
The Measure of Audio, AES UK Conference 1997.

later on Frindle mentioned differences detectable as low as 4µs.

I have that paper. He lists a bunch of things that he claims are audible, but neither references the tests nor (for the ones he did) gives any detail whatever on how the tests were done and what precisely was being tested. It's an entertaining document, much of it is useful, but it's rather short on data. I wouldn't be citing it as evidence that a 10us interchannel time delay is audible on speakers.

I do note with amusement that he says several times that the quality of the equipment used in evaluations isn't very important as long as level and frequency response variations are controlled.
 
Hi,



That is completely besides the point.

So you insist on not showing anything real that relates to how things really work while presenting something that is an idealised version of hos things could be done, but are not. That is your prerogative...

Ciao T


You insist on some claims that have no ground in theory and in practice.
My (and many others) measurements and listening tests on real equipment correlate very well with the theory so there is no gap between the two world.

Also all the measurements and listening tests done by Dan Lavry and Paul Frindle are not some theoretical simulations but results with real converters and processors.

here is a small excerpt from Dan's paper:
"Dithered signals provide a "constant" noise floor, independent from signal and DC offset.
Measuring a triangular dither with a "dB meter" shows a reading of -93 dB for 16 bits. The energy
density (at each frequency) is at about -125 dB. Can you hear under the noise floor? You can hear
30dB below your "meter", all the way down to the noise density in the surrounding frequencies.
Reexamination of the frequency plots shows that you can hear a 16 bits dithered signal down to about
-125 dB under full scale.
Some manufacturers choose to view the special case of undithered signal gating as a the 16 bits
hearing threshold. One should not confuse the "gating threshold" of unditherd system with the noise
density of a dithered one. The "special gating case with 1/2 LSB of DC" occurs at about -96 dB. The
noise density (per frequency) for dithered signals is almost 30 dB lower.
The ability of the dither "to to bring the gated signal back" is shared by all types of dither.
Rectangular, Nyquist and triangular all perform the task within about 3 dB of each other (a range of
about 1/2 a bit). Beware of claims for "a special ability" of a specific type of dither to provide "3 -4
more bits". The 30 dB or so of dynamic range "beyond" the gating threshold is not unique to one type
of dither. It is shared by all types of dither and is not to be confused for additional bits. The proper
criteria for dither quality is its ability to eliminate distortions and noise modulation."


chrissugar
 
Is it? Would you say that applying 14 Bits of worth of dither and converting a 24 Bit PCM Signal to 10 Bit with this is a good idea?

If the final delivery format is 10bit then yes but why we are talking about a limited format.


You are aware of Ethan Wienders tests?

It is Ethan Winer and yes I'm aware for more than 10 years about all of his demistification. His tests are mostly based on mediocre quality recordings. I'm not surprised he thinks the Creative Sound Blaster is as good as an Apogee converter.
Nobody in the mastering comunity is taking him serious.
I think you should listen to some of the guys who proved their hearing and technical abilities with their work like Keith Johnson, Dan Lavry, Paul Frindle, Bob Katz and many others.


Funnily enough, that is not what I hear and what for example Gremkin/Wiener found.

I think you are alone because the whole mastering community (you know the guys who deliver all the audiophile music available) think different related to dither.
Dither is not an option in the real world audio.

chrissugar
 
You refer to images, plural, and signals, plural; then later refer to sources. My impression is that we started on the topic of stereo DAC, then it was suggested that a 10 us delay could be heard. It seems that most of the replies have been in regard to stereo, such that when you delay the right channel with respect to the left, there are no "other" signals. The right channel is the only other signal compared to the reference, as in singular, not plural.

First, a 10 uSec interchannel broadspectrum delay is easily audible..trivial. But it can only be discerned using headphones. Attempts to counter the image shift using a pan pot has mixed results, as all the image does not shift exactly the same, it is frequency dependent. Some of the content appears to shift easily, some of the content "locks" off axis, and panning necessary to center the locked content shifts other content too much. To hear this, you must use a mono signal of course. Note: this is my personal observations over about 4 years, and of course requires duplication by others.

A 10 uSec interchannel broadspectrum delay in speakers is NOT audible. Humans will simply adjust to the sweetspot shift.

When a SPECIFIC voice or instrument is shifted 10 uSec interchannel with respect to other content which has not been shifted, humans will perceive the drift of the shifted content RELATIVE TO the unshifted.

If you want to talk about multi-mono mixing consoles, pan laws, binaural recording, multi-microphone techniques, and surround sound reproduction, then I think you're opening an entirely unrelated set of topics. Have mercy on this thread!
Who's kidding who?? This thread is already like that...:eek:

Cheers, jn
 
It is Ethan Winer and yes I'm aware for more than 10 years about all of his demistification. His tests are mostly based on mediocre quality recordings.


Read down a way... Looks very trustworthy.

"The hum thing is a little disturbing. The SB was looking for a -10 dBu level (I assume) and the Apogee was looking for a +4 dBu level. We cranked the Apogee input "trim" all the way up to make the signal even with the SB input. In retrospect, I should have adjusted a dip switch for -10 nominal level operation which would have required less cranking up the gain. Other explanations could be the path between 1202 and Apogee. I know I've never made a balanced recording with it and had any audible hum.

By the way, hum or no hum, I don't feel this really hurts any validity of what we did. I don't think 60 cycles and its harmonics will affect at all the way something like a triangle will be recorded. And I doubt there was any significant impact on the guitar track either. Of course we traded off running balanced for doing a split to get two simultaneous recordings, a fact that is very important to the whole experiment. "
 
Hi,

If the final delivery format is 10bit then yes but why we are talking about a limited format.

Practically all modern ADC and DAC's operate preciely like this, even if the writing on the outside says "23 Bit" or even "32 Bit"

I think you should listen to some of the guys who proved their hearing and technical abilities with their work like Keith Johnson, Dan Lavry, Paul Frindle, Bob Katz and many others.

Actually, there are reasons why Keith Johnson uses mostly his own designs, even for converters. As to the rest, I am familiar with them. I said it before and will say it again, just because X or Y says so does not make it true.

Ciao T
 
Scott,

What I presented was text book stuff.

What you presented had NIL to do with how things are applied in reality. Of course you DO NOT HAVE TO show things that bear any relation to reality.

I would think converters used to master CD's are a bit better than off the shelf and their INL and DNL are pretty close to ideal, in any case things are much better now.

I would think I personally would look up some of the real stuff and check my assumptions in the harsh light of reality, but that's just me... Others prefer to simply AssUMe...

If you sent a complete map of the INL and DNL of your particular A/D or D/A it could be simulated, but no I don't have time.

No need to that. Instead show the Dither that is USUALLY applied and not you arguably clever and "a good idea" subtractive version. No need to include the converters as "non-ideal", it is not required for the demonstration.

As for the harping on FFT's, ideal TPDF dither produces uniformly distributed white noise as a residual, the spectra are not interesting on any scale. Text book stuff again.

A squid eating dough in a polyethylene bag is fast and bulbous. Got me?

Ciao T
 
OK, so my "non-sequiturs" led me to the same conclusion as yours. :D

Can you suggest a signal where 10us delay in one channel for headphones is "easily audible.... trivial?" I'd like to try it for myself.

Wooly Bully by Sam the Sham and the Pharoes(sp)
Eli's Coming Three Dog Knight.
Cheryl Lynn To be Real.
Inna Gadda Da Vida Iron Butterfly
Thompson Twins In the name of love.
CJ and Company Devils Gun..

Anything you want.


You may spend days and days reversing headphones, reversing channels electrically, altering the panning without looking...as long as it takes. I've done it for years. Constantly testing my results every chance I get...

You set the parameters, make yourself happy that you are not just imagining anything..you know, standard protocol..

Cheers, jn

ps. The verbage "non sequitur" was because I specifically stated "with respect to non shifted content", or sumptin like that.
 
Hi,

32 bit converters? What kind of connector does that use?

If we are talking about the chip, I2S which with a BCK of 64fs is in effects a 32 bit format. In terms of a system, USB and PCI based Audio can work at 32 Bit and even 384KHz sample rate if needs must.

Of course, most so-called "32 Bit" Dac's and Adc's are actually 8-12 Bit parts with dither and noiseshaping giving at best around 21 Bits ENOB with A-weighting and looking non too closely, only one current part manages actually 22Bit ENOB as DAC. But it still says "32 Bit" on the package...

Ciao T
 
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