John Curl's Blowtorch preamplifier part II

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Sorry to press the question (you guys wander all over the place ;) ) can we agree that with a good 16 bit system, with 118 dB SNR, it is not useful to have a 24 bit reproduction system, as the reproduction system won't make 118 dB SNR anyway? If not, why not?

Jan
As I said before, almost every 16 bit (usually 44.1/48kHz) reproduction system manipulates the signal one more time, before it gets converted back to analog. Such system contains some form of DSP (!) - digital filter, upsampler etc.
With 16 bit input that DSP has less 'information' to calculate its output than with 24 bit input and might introduce yet more errors.
DSP/math guys, please correct me if I'm wrong.
 
DAC outputs same level whether in pro gear or domestic. Feed to speakers is same level unless there is a significant spl or efficiency difference. So if the total gain is the same I can't see a snr improvement. Running interconnects hot may have some theoretical benefits, but if you run balanced in the home you can get a noise free digital playback system to any sane measure.
See you are looking at two very different venues home short distances well shielded balanced cable low field emf etc . Pro audio long distances sometimes 100 meters plus with lots of lights using real voltage and current often a fair number of kw of lights with scr controllers generating a lot of noise broadcast over a medium spectrum of rf . So what is need in pro audio to over come a noisy rf field does not happen and thus not need to be used at home. And by running the line hot if you lose a bit of level down the line you still have lots to work with .Hope that helps.
 
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The internet speed is there, the cpu and DSP speed is there, the memory storage capacity is all there now at mainstream affordable costs. So now is the right time to start using 24b/96k exclusively and someday, down the road, 32 bit downloads.
The only ones left to get on side, are fools :spin: and the ones owning/distributing/profiting from the Intellectual Property.
Think about it, their IP, if it is on tape, has a limited life span, so a wise investor would want to preserve their IP forever. Minimal investment for your profitable IP. You being the investor, would you not want to preserve your IP in the best way possible? If your source IP is already hi-res digital, it is a BIG marketing game to distribute the IP in low res. Just so that you think one day that you will get the sucker customer to re purchase the IP again, so one has all the flavours, just like candies. = ridiculous greedy world.
So that is, Richard, me, Neil Young being on side, for hi-res audio distribution, at affordable prices, who else ...
Well I could just digitiize some of my fav LP's at 24/96, this time, but I better get a ultrasonic LP cleaner going first.
I request >0.1% thd+n(Sinad) at -60dBc, does 24b/96k get me to this target? or do we need some more/newer technology to do it?
 
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See you are looking at two very different venues home short distances well shielded balanced cable low field emf etc . Pro audio long distances sometimes 100 meters plus with lots of lights using real voltage and current often a fair number of kw of lights with scr controllers generating a lot of noise broadcast over a medium spectrum of rf . So what is need in pro audio to over come a noisy rf field does not happen and thus not need to be used at home. And by running the line hot if you lose a bit of level down the line you still have lots to work with .Hope that helps.

Ok should have added the caveat that I was talking merely about RNM saying that domestic should use pro (as in studio) levels. For sound reinforcement I understand. I didn't want to get into the balanced vs single ended in domestic environment, just making the point that the 0dbFS=2V that good domestic digital equipment puts out does not put you at any disadvantage in the home.
 
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A zero and cal cycle get you a lot, after all they make very accurate digital scales legal for trade with far more than 12bits resolution.

For the ADS1232, the reported nonlinearity is around -104dB rel input.
Not very low for a 24bit converter but weighing scales -and measuring instruments in general industry-are tested and certified as complete systems, not components.
Thus, although an analog core unit may have excellent linearity figures, a complex digital measuring system based on ADC, far outperforms the system based on the analog core unit in terms of repeatability, reliability, noise immunity, short & long term stability (think of a wide body aircraft weighing system, or of laboratory grade calibrators).

I don't know of published time stability data, but we do take mass quantities of it in characterization.

I remember from reviewing the cal cert correction factors from ADC based calibrators like Fluke 5080, Ruska 7250 ect that the annual drift was minimal and always within the predicted (spec'ed) range.

I'm not sure what to say about the different ways of specifying it, and measuring it gets very complicated when one worries about things like how many codes are exercised.

It seem that code oriented testing is a hard way to quantise such measurements, the traditional way is closer to our understanding (more intuitive).
I am aware of the definite turn of treating 'testing' as 'measurement' (metrology territory)

Thanks for responding

George
 
I don't tolerate distortion but i don't listen at very high volume and my system is quite respectable so i don't hear any unless the recording contains it. Some rock bands deliberately include distortion in their material so seeking to remove that would produce a false outcome.
Well, I do! The member with DEQX setup, the first time I listened to it he had it on a very low level, so I asked him to up and up it - pretty soon we had reached the limits of adjustment - he has adjusted the gain chain so that that no clipping could occur at any time. The point being, that when you hear at realistic levels then the distortion artifacts are extremely obvious, and then if you start to drop the volume it's relatively easy to follow the sound of that "wrongness" still occurring. As an example of how one diagnoses, if that wrongness goes away at some point of steadily attenuating then it is most likely power supply related, or also possibly interference from mains current spikiness affecting other components.

Maybe i should have asked if you could give us some examples of what you found wrong in a particular setup and how you rectified it.

I found that well designed circuit layout is crucial, then decent components with closer tolerances can provide some improvement. Recently i moved my power amps close to my speakers so i could use short speaker cables, my buffer (DCB1) is easily able to drive 15 foot interconnects, that brought some benefits too.
Most times I hear the treble being wrong, it is just not convincing, it has a turgid, irksome, "hifi" quality about it - you can't increase the volume without that aspect becoming impossible to live with for any decent period of time. And unfortunately, nearly all the problems that beset audio setups have most impact in that area. As an extremely downmarket example of the process, my PC speakers - that people here love to laugh at, ;) - had typical grotty sound to start with; but it also essentially had nothing glaringly wrong with it - a very promising sign! And the major issues were the usual suspects - cheap hardware in key areas, like connectors and potentiometers. So I bypassed all of these, every last one of them, and then gave the speakers a good thrashing - the downside of cheaper gear is that they always need to be conditioned heavily to reach a stable, and optimum state. Part of the story here is that the capacitors are pretty low grade, and need to be exercised strongly to perform well. Last step was to optimise the power feed to the computer and speaker internal amps, to minimise interference from this factor.

Doing this was sufficient for these little fellas to give a good account of themselves - a lot of material could be run at max volume without obvious problems - power supplies are still the biggest issue, the "wrong" material makes this obvious; but it's not worth putting in more effort, considering the overall balance of the situation.

As for cars, i used to be a mechanic back in the 1980's before i became a marine engineer. Believe me i've heard some crackers. It's making a 'going round' noise etc. A test drive usually helped, also i had a quite sophisticated Crypton engine tuner at my disposal which i was trained to use. Cause and effect Frank, just like in audio.

One more thing, if it ain't broke, don't fix it :bulb:
Trouble is, nearly all systems are broke - especially the expensive ones!! A good dose of fixin' is exactly what they do need - the thing is to be able to "hear" the problems, which you may, or may not, be able to do something about ...
 
Along the way, thinking minds can tell why analog has always sounded better..... near infinite resolution limited only by distortion and noise--- both of which are extremely low over the range of levels we listen. The only problem is how to get direct feed sources in analog -- from master tapes or a live broadcast (FM). Only the live broadcast is practical but all too rare.

Digital can be very close to analog IF we get the bits up and the sampling rate up higher. 24/96 is VERY good. And, yes, it will mostly be done in IC's.



THx for listening--RNMarsh
The thinking's all wrong here, Richard - yet again, :D !! When digital doesn't sound good it's because something is not working right in that system. Don't chase goodness - eliminate badness!

When I first got brilliant sound, from "miserable" CD, :p - I went on a mad binge, checking out every type of system I could get a listen to, to see if that same quality would be replicated. With CD this never happened, they were uniformly either dreadful, or discarded a huge chunk of the sound to make themselves "nice" - in analogue it was all over the place, only a premium LP setup with Goldmund Reference TT, etc, made me say, aha!! ... you can hit the mark with vinyl too ..
 
Wouldn't dream of it. Love going there, but you have to admit it's not the best shape. Proms only moved there when the preferred venue was somewhat redesigned in the 1940s.

I do agree, but after 45 years of Proms I've come to love its bouncy sound, better of course after the ceiling mushrooms (1965?) were added. I have found an area under a higher tier where the orchestral sound is very pleasing. To me.
 
Another key point is: unless you're prepared to get fussy, really fussy with digital replay ... it ain't gonna happen - convincing sound, that is. Being matter of fact, or casual about it will almost certainly guarantee disappointing sound - at the moment typical replay environments are extremely fragile, the slightest "wrongness" somewhere will knock the subjective quality of digital sound right off its pedestal, instantly ... I'm afraid that is just the nature of the beast, at the moment ...
 
As I said before, almost every 16 bit (usually 44.1/48kHz) reproduction system manipulates the signal one more time, before it gets converted back to analog. Such system contains some form of DSP (!) - digital filter, upsampler etc.
With 16 bit input that DSP has less 'information' to calculate its output than with 24 bit input and might introduce yet more errors.
DSP/math guys, please correct me if I'm wrong.

There are plenty of playback devices that can be set to basically send the data directly to a DAC without further processing. There are plenty of threads around concerning bit perfect playback.
 
So now is the right time to start using 24b exclusively and someday, down the road, 32 bit downloads.

IMO.


THx-RNMarsh

Integer or floating point? 32 bit integers are an extra 8 bits of noise and 32 bit floats are 24 bit mantissa and can be arbitrarily normalized and maintain the 24 bit accuracy so they are computationally useful in DSP's but don't make sense as a delivery format.
 
I take it that almost nobody has done the exercises of starting with normal CD, upsampling to hires - or starting with premium hires, downsampling that to CD quality, then upsampling that result back up to the starting hires format, and comparing them all ... all the simple, obvious things to try first ... oh dear, :rolleyes: , :rolleyes: ...
 
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