John Curl's Blowtorch preamplifier part II

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Bear if you don't want to do the homework and become facile in the theory but on the other hand throw out random criticism there's no basis for a discussion.

Best learn the meaning of the word criticism, Scott.

I've only asked a question based on an idea - perhaps my idea is ill formed, I am not sufficiently informed, or anything like that. But thus far, in areas where I am reasonably well informed, folks like you have repeatedly balked when asked to take a stand - just read back.

Or do you want to say that all "blameless" amplifiers sound the same?
Scott, do all DACs that meet the mfrs suggested topology or better sound the same??
How about it?
SY?

C'mon down off the high and mighty thrones.
you know but ya ain't sayin??



The bottom line is simple enough, digital is not perfect, not perfectly transparent in all regards. Nor is analog. The question is WHY.

IF one gets back seemingly divergent results, in particular that it is possible to reform a sine wave at 1kHz essentially without measured distortion, but if the slope/rise time of that sine signal is increased (approaching square wave within the Nyquist limit for the system), and then there is a change in the ability of the method utilized to recreate the signal, especially when compared with the same bandwidth analog reproduction of the same signal, then that is a difference.

There is a difference between the two results...

Where should we look, if one does not think that digital is "perfect sound forever"?? It doesn't seem to be in the measurements and specs that are presented, does it??
 
Best learn the meaning of the word criticism, Scott.

I've only asked a question based on an idea - perhaps my idea is ill formed, I am not sufficiently informed, or anything like that. But thus far, in areas where I am reasonably well informed, folks like you have repeatedly balked when asked to take a stand - just read back.

Or do you want to say that all "blameless" amplifiers sound the same?
Scott, do all DACs that meet the mfrs suggested topology or better sound the same??
How about it?
SY?

C'mon down off the high and mighty thrones.
you know but ya ain't sayin??



The bottom line is simple enough, digital is not perfect, not perfectly transparent in all regards. Nor is analog. The question is WHY.

IF one gets back seemingly divergent results, in particular that it is possible to reform a sine wave at 1kHz essentially without measured distortion, but if the slope/rise time of that sine signal is increased (approaching square wave within the Nyquist limit for the system), and then there is a change in the ability of the method utilized to recreate the signal, especially when compared with the same bandwidth analog reproduction of the same signal, then that is a difference.

There is a difference between the two results...

Where should we look, if one does not think that digital is "perfect sound forever"?? It doesn't seem to be in the measurements and specs that are presented, does it??

They have been patient, but you have fundamental misunderstandings on how this all works. That leads to the same old, tired, and flawed arguments that get recycled here every few months.


"approaching square wave within the Nyquist limit for the system" - you can't approach a square wave and still be within the Nyquist limit of the system.

Create a band-limited analog system with similar cutoffs and slopes and let me know what you find.

Everything agrees with the math.

Could be that we're closer to perfect sound forever on the electronics side than certain people in the industry would like you to believe. There are probably a lot of people that think all decent equipment sounds pretty much the same, but when you regularly visit echo chambers filled with people carrying pitchforks you aren't going to get many dissenting opinions.
 
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Before showing up at a listening test, it would seem reasonable to agree to some specifications as to how it will work, what is acceptable equipment, etc.

When I compare sounds, I normally use near field speakers at low to moderate volume in a quiet room. I am free to move around at listen at different distances and angles, and there is no grill cloth on the speakers, nor any similar obstruction between me and the speakers. Sometimes I like to use headphones. Probably such requirements would not be a problem for DACs or amps. But I wouldn't want to show up and find out I have to sit in a chair a few feet away from a speaker behind a curtain, and with people walking in and out, talking, etc.

Also, I know with some speakers of good reputation for general listening I can't even hear very well if a mix has certain types of flaws. If I show up and there is a nice sounding speaker with a nice very flat frequency response, but it it also happens to obscure some features of the sound not related to frequency response, that could be a problem for getting accurate results.

That's why in the past I have said someone can come over to my place and listen here. I haven't found anybody that can't learn to hear things that are actually there given some training. My experience so far is that anyone can hear the differences between my DACs, given my amp and speakers, for example, once it has been explained what to listen for.
 
They have been patient, but you have fundamental misunderstandings on how this all works. That leads to the same old, tired, and flawed arguments that get recycled here every few months.


"approaching square wave within the Nyquist limit for the system" - you can't approach a square wave and still be within the Nyquist limit of the system.

Create a band-limited analog system with similar cutoffs and slopes and let me know what you find.

Everything agrees with the math.

<snip>

Ok, fine - but which filter (aka approximation) is "right"?

I understand that reproducing a clean 1kHz square wave in the bandwidth limited system that is 44.1 is not possible. Something is reconstructed that best resembles the original.

By using different filters, you get a different waveform approximation out.
Between any two filters by definition, there is a difference.

The difference between the original and the output is "error".
Upfront all I suggested is that this "reconstruction" in effect is done twice, not once with digital recording. Might this be a source of increased error? That's the premise.

Or put another way, how many times passed through assorted similar but different FIR filters of the same bandwidth & similar slope can you go before you start to see any anomalies. Or does the signal when finally reconstructed after the "last" filter still resemble the original source as much as it did through any of the filters individually?

If in fact you fellows know as much as you pretend, this ought to be a very straightforward thing to reply to.
 
Seeburg 1000 Background Music Radio | seeburg1000.com Click the Playing Now and tell me how long you can take it.

It's having a nice salutary effect on my blood pressure as I parse through all the folks that enjoy violating sampling theory/physics/etc. :D

Dunno how much past 20 minutes I'll last, but if I were doing an activity that had the music a bit more background-y, I'd imagine I could go a long time. Of course, I have ambient-ish electronica like Emancipator, (e.g., https://www.youtube.com/watch?v=k7Z7USWo2Lk), coming through my headphones pretty regularly.
 
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Haha, gotta say I can't do Christmas music too much; most of it drives me absolutely nuts.

Then again there are some absolutely amazing choral arrangements that I very thoroughly enjoy (and, IMO, sound marvelous regardless one's religious affiliation). Especially live with a group of likeminded friends.
 
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To whit:

From 2016 reviews in Stereopile mag. (randomly found)

Would anyone here accept an amp that passed a square wave that looked anything like this??

Those look like 1 bit audio showing what you get at -96 dB for a 16 bit file. They really aren't bad. And I defy you to hear that tone after setting a -20 dB reference to 80 dB spl at your listening position.

I'll show you some square waves from real DACs at real audio levels tomorrow. They do not look like your pictures. (I know all about 20 KHz square waves from power amps. . . )

The idea of complementary filters is not new. Its part of the HDCD spec and curiously now the MQA spec. For a 192 KHz file its probably not an issue, for a 44.1 file it may be.
 
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Oh you guys would HATE coming over to my house. :D This is what I've been grooving to for the past couple of days. It makes me happy.

Seeburg 1000 Background Music Radio | seeburg1000.com Click the Playing Now and tell me how long you can take it.

I think it's quite nice. Airy highs, good bass extension, and clearly on the piece I just listened to, very good musicians.

Thanks for sharing!

More l8r ;)
 
Yeah, I need to keep reminding myself not to respond to anything that neither is sensible nor honest.

You are constantly using the misnomer "ears only" although you know that it is a misleading phrase (or better "ears only" is simply not true), which is dishonest behaviour, hence you shouldn´t imo speculate about others possible dishonesty.

Oh my, is someone constructing signals that violate Shannon-Nyquist and wondering why things end up badly?

(Just because you made it in software rather than lowpass-ADC, doesn't mean that aliasing isn't going to be a big problem)

But right from the beginning of the CD age we were using measurement signals that were carefully constructed in the digital domain and contained in the measurement disc sets from Matsushita and the like.

Mentioning software points to another problem; it isn´t ensured that all those software packages people are using to create music or process music only produce signals that look like as if they were done in the analog domain and then digitized.
In other words you will most likely find signals on cds that you could never have produced with proper band limiting in front of an ADC.
 
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