John Curl's Blowtorch preamplifier part II

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When I search the AES library for "Dither" I get:

Authors: Blessener, Barry; Lee, Francis F.
Affiliation: Massachusetts Institute of Technology, Cambridge, MA
AES Convention:40 (April 1971)

Author: Blesser, Barry A.
Affiliation: Blesser Associates, Raymond, NH
JAES Volume 26 Issue 10 pp. 739-771; October 1978

and!

A dynamic range of 118 dB is determined necessary for subjective noise-free reproduction of music in a dithered digital audio recorder. Maximum peak sound levels in music are compared to the minimum discernible level of white noise in a quiet listening situation. Microphone noise limitations, monitoring loudspeaker capabilities, and performance environment noise levels are also considered.

Author: Fielder, Louis D.
Affiliation: Ampex Corporation, Redwood City, CA
AES Convention:69 (May 1981)
 
No. No dithering helps you to get under 1/2 LSB in A/D conversion. Just to add I am interested in transient signals digitizing.

Pavel,

A transient is in theory of infinite bandwidth, therefore after the LPF filter should show up as the maximum frequency component passed and as mentioned jitter (or ratiometric system process gain) decreases as frequency increases. So we agree at maximum frequency dither does not provide enhancement.

One other issue mentioned was a Vanderkooy paper which if I recall it correctly did a long math treatment but ignored the physical implementation. If you have a sample and hold grab your input signal, how can you tell what the A/D conversion process is?

Finally I started by using a 16 bit Discrete Level and Discrete time sampling system as being too limited for my idea of audio use. Virtually everyone assume I was talking about CD use. Since most of my work is on live use, that is not what the image in my mind was.

So there are still quite a few translation issues to be worked out.

SY he is just plain wrong! :)
 
Works in the other direction as well. Spectra showing signals coming out of D/A and anti-imaging at well under 1/2 LSB have been posted here repeatedly, apparently to no effect.

Show me how it works in A/D, to get below 1/2 LSB resolution. How you get the right digital representation of input signal of right amplitude at right time, with resolution higher than 1/2 LSB.

You need exact and precise A/D conversion to get the signal for CD playback.

I am not interested in something periodic, delayed, averaged and placed at wrong time at D/A output. No empty words, but facts please (as you like it).
 
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Show me how it works in A/D, to get below 1/2 LSB resolution. How you get the right digital representation of input signal of right amplitude at right time, with resolution higher than 1/2 LSB.

You need exact and precise A/D conversion to get the signal for CD playback.

I am not interested in something periodic, delayed, averaged and placed at wrong time at D/A output. No empty words, but facts please (as you like it).


So basically, you want to violate Nyquist?

Music is a flow of transients, not steady state periodic sinuses.

Much more like the latter than the former. At least for music that uses notes and tone.
 
Simon, sure, I would correct it as a non-periodic signal limited in bandwidth. I do not want to speak about infinity etc. I am sure you know what I mean.

Music is a flow of transients, not steady state periodic sinuses.

Pavel,

I think we are having trouble communicating. I can design a waveform that sounds like noise but will allow low frequency components of the signal to bounce up and down enough from their nominal value that they will get multiple encodings over their period. When reproduced (as signal and out of critical band noise) this is almost the same as doing more step sizes to increase the resolution. It does not require the input sample to be periodic, only to be long enough to get multiple samples over the base period.

ES
 
This is getting out of hand. Scott, I sat in at AES on at least one of Barry Blesser's papers, probably 1978. What do you think I did, on those days, for a living?

Not digital, Maybe Barry was not in the inner circle and losing SNR below the 16 bit level was politically incorrect to the "perfect sound forever" crew. When the raw resolution of A/D's got better the issue is less important.
I have the AES digest from 1982, dither was a very small footnote. Serious evaluation of different methods had not been done. Saying digital has problems and dither won't fix it is not a particularly useful input without even any data.
 
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1982 AES Premier Conference on Digital Audio Barry Blesser is editor, his kickoff paper on the basic aspects of digital audio does not mention dither. It is mentioned in passing later noting some of the problems with his add/subtract method. It is not mentioned anywhere else I could find. It was clearly very much in its infancy in 1980 and I don't see how any definative discussions on techniques for appying it were going on, perhaps except to discredit digital audio.

??? I am completely puzzled. I quoted this:

Blesser investigated quantization noise without
mention of dither in an earlier paper [8], but in his
comprehensive examination [9] of digital audio he
clearly indicates the beneficial effects of dither
and
points out that the average value of the quantized signal
can move continuously between two levels. It seems
clear from his descriptions that for digital audio the
concept of dither does not necessarily imply the addition-
subtraction scheme of Roberts [3], but simply
a noise added to the ADC input to eliminate digital
artifacts.

And this, which is referenced above:

[8) B. A. Blesser, "An Investigation ofQuantization
Noise," J. Audio Eng. Soc. (Project Notes), vol. 22,
pp. 20-22 (1974 Jan./Feb.).
[9] B. A. Blesser, "Digitization of Audio: A Comprehensive
Examination of Theory, lmplementation,
and Current Practice," J. Audio Eng. Soc., vol. 26,
pp. 739-771 (1978 Oct).

It was a JAES Paper from 1978! Not 1982.
Why are you insisting that he did not mention dither??
 
Hello Thorsten

>>Having tried it I prefer to do the EQ in analog hardware and to convert to Digital afterwards, strictly for subjective sonic reasons. But I also like multibit DA's without dither and followed by tubes that produce a completely scandalous around 0.2% 2nd HD at digital full scale, so WTFDIK?

When you refer to digital full scale what is the RMS level of the analog output signal.

Regards
Arthur
 
??? I am completely puzzled. I quoted this:
And this, which is referenced above:
It was a JAES Paper from 1978! Not 1982.
Why are you insisting that he did not mention dither??

Did you read what I said? He had a small mention of dither in a later paper of the same conference. His keynote paper did not mention it as essential (or at all). Even in 1982 it seems the industry (AES) as a whole did not consider it very important. In fact Barry discovered it at MIT and didn't bother publishing it, his technique as Lipshitz mentions was apparently never developed further. Apparently his 1978 paper did not light any fires. Ed found only 3 papers pre 1982, but I don't know if he was exhaustive. Besides dither is so close to things like stocastic resonance that I don't believe there is not a lot more relevant papers in other disciplines. So forget it, I have no idea how the AES decides what's important

EDIT - Ed found a mention in 1971, I find it strange it wasn't studied fully well before 1981.
 
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Correct, it fills the analog equaivalent of the lower bits with "fuzzy distortion", which depending on specific implementation may be semi-deterministic, signal dependent or fully random. So there are no longer discrete levels, mainly because if we WERE to see the discrete levels of the converter we would realise that our 32 Bit converter is actually only a 12 Bit one.

In fact, apply dither to a sharp transient and you can literally SEE the fuzzy distortion on any decent 'scope.

What *is* this fuzzy distortion, later quoted to be 50% or 1000%, that isn't measurable, and must be at least pretty hard to hear (since most people can't hear *any* difference in many serial passes through an ordinary stock 44.1/16 bit A/D/A)? Is it some kind of modulation of signal, or are we talking about simple noise? Or what?

The word "transient" if fraught with peril - A/D/A conversions are necessarily bandlimited. Could you suggest a setup where I could see this fuzzy distortion for myself? (Got lotsa scopes.)

Thanks,
Chris
 
What *is* this fuzzy distortion, later quoted to be 50% or 1000%, that isn't measurable, and must be at least pretty hard to hear (since most people can't hear *any* difference in many serial passes through an ordinary stock 44.1/16 bit A/D/A)? Is it some kind of modulation of signal, or are we talking about simple noise? Or what?

The word "transient" if fraught with peril - A/D/A conversions are necessarily bandlimited. Could you suggest a setup where I could see this fuzzy distortion for myself? (Got lotsa scopes.)

Thanks,
Chris

In my simulations it looks like noise, I suppose added noise is a fuzzy distortion. I don't see a controversy here. A simple truncation experiment works. Take a floating point 1k sine wave at -80dB make two copies and add -80dB rms gaussian noise to one (not ideal but proves a point) and tuncate them to 16bits with no dither. The one with added noise does not have the THD and the other looks exactly like the input fed through an ideal simulated A/D. So the official CBS/SONY CD test disk (circa 1983) was incapable of testing the true low level performance of a DAC since the test waveforms are packed with in band tones/distortion that could have been eliminated with dither.
 
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