John Curl's Blowtorch preamplifier part II

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When I get masters of 24/96, and compare to CD/16b, the 24/96 recording can be recorded further down and away from clipping -- a common problem for recordest. If I need to be at least -20 down to avoid clipping of pop music or -30 down for other types of music, it brings me closer to the noise floor. With a 90dB dynamic range on 16b that will put me close enough to the noise that it can be heard. But with a 24/96 recording, I have so much dynamic range that i can record at much lower levels and still be far from the noise floor.

You have any examples to back this up? Running 24/96 tracks I have though the DR plugin on foobar shows peak levels around 0 to -1dBFS just like the 16 bit masters and the RMS levels likewise (or even less like the new zep releases). Very little is recorded with DR over 20.
 
When I get masters of 24/96, and compare to CD/16b, the 24/96 recording can be recorded further down and away from clipping -- a common problem for recordest. If I need to be at least -20 down to avoid clipping of pop music or -30 down for other types of music, it brings me closer to the noise floor. With a 90dB dynamic range on 16b that will put me close enough to the noise that it can be heard. But with a 24/96 recording, I have so much dynamic range that i can record at much lower levels and still be far from the noise floor.
I have a CD which has been mastered at a ridiculously low level, The Essential Odetta - Wikipedia, the free encyclopedia, the label and recording year gives an idea of what the nominal "quality" should be - at maximum volume this is about as loud as a kitchen radio in the morning. One of my more valuable test CDs, in the quietness of her lone voice I can hear the large space she's performing in, and all the subtle vocal inflexions and minute non-musical scrapings on her guitar - there is nothing in the recorded sound then that betrays its digital source ... sorry, the dynamic range argument is a nonsense ...

Clipping is so common in recording that there actually exists software that will put the peaks back on a flat top/clip waveform. Thats an interesting trick to enhance effectively dynamic range while staying further from the noise floor. Dont know what that does for the distortion caused, though.

THx_RNMarsh
This can be done very intelligently, I've done a number of rounds with the best of what's readily available - and it is very effective. Even knowing exactly where the damage was originally done, I have found it impossible to pick an audible artifact after the "fixing" ...
 
So why say SMPS is 'asking for trouble' then backtrack? Bad examples of all power supply technology exists.
Because the companies that typically will readily use a SMPS because it makes good engineering sense often are also the ones that don't do the right sort of critical evaluation of the audible effects. I heard a DEQX unit being demonstrated many years ago, that would have used that original SMPS supply - typical, competent sound, but nothing special - OK, you can straighten out the FR of that speaker "perfectly", but there is nothing in the sound that makes me say, Wow, that's so much better!

But the modified DEQX I heard recently was able to pull off high quality sound - not all the time, depending as always, but it definitely was capable of delivering the "specialness" that I always look for.
 
The "good engineers" "listen" to their measuring equipment - that's their focus, :p. They're wanting to achieve certain technical levels of performance, which takes a fair bit of effort - perhaps nothing left over to then refine the subjectively apparent "rough edges", at least in the first version or two.

This is what has happened to the DEQX - each iteration has improved the unit, the reviews note "that it now sounds better" - yet technically nothing radical is happening to cause this.
 
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there is nothing in the recorded sound then that betrays its digital source ... sorry, the dynamic range argument is a nonsense ...

This can be done very intelligently, I've done a number of rounds with the best of what's readily available - and it is very effective. Even knowing exactly where the damage was originally done, I have found it impossible to pick an audible artifact after the "fixing" ...

Besides my own recording experience, I get a lot of my recording info from the known Master recording engineers themselves and what they have said/written. So, excuse me, I am not impressed with your explanation.

It is good to know you can clip and then fix it so it would sound just like the unclipped version with such a fix. :rolleyes: I prefer the sound of the unclipped and Not fixed later approach to recording.


THx-RNMarsh
 
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The "good engineers" "listen" to their measuring equipment - that's their focus, :p. They're wanting to achieve certain technical levels of performance, which takes a fair bit of effort - perhaps nothing left over to then refine the subjectively apparent "rough edges", at least in the first version or two.

This is what has happened to the DEQX - each iteration has improved the unit, the reviews note "that it now sounds better" - yet technically nothing radical is happening to cause this.

Some would ask how do you know -- did you do a carefully setup DBLT between un mod and a modified unit? What are the T&M data to also back those subjective claims? ETC. I get that all the time... mostly from SY. What do you say to that argument?


THx-RNMarsh
 
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When designing an instrument to meet a certain measure of performance it is nice to not have a moving target. So Charles Hansen's open-loop Pono IV wins praise in some quarters I'm sure. OTOH...

So what's the target? The numbers, the right colored sound? BTW dvv don't sell yourself short I'm sure you can make some very credible designs that sound fine even with a bunch of 8-legs

For me? The target is the lowest possible distortion under normally expected loading. That would be THD minimum of -100dB re 1v or more. Therefore the -5 is my choise. I want all pieces to be at least that low so the total in the whole record thru playback chain remains below audibility.

You designed a line stage with much better than -100db with high voltage bipolar power supply (-120?) direct-coupled into 600 Ohms for DIYaudio.... that is a good way to go IMO. Old school... no added distortion and noise to the SOTA possible. But, that is just where i am coming from.


THx-RNMarsh
 
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Besides my own recording experience, I get a lot of my recording info from the known Master recording engineers themselves and what they have said/written. So, excuse me, I am not impressed with your explanation.
Well, it's not an explanation - just personal experience, :D. I've done a couple of rounds of deliberately, dramatically making it much worse for Red book recordings, getting them to misbehave, audibly, by throwing away bits of resolution - trying to force digital to "sound bad" - I get to hear that the DAC in one channel is a bit noisier than the other, say, by going to such extremes, but as far as listening to a conventional recording at any reasonable volume, and hearing more "distortion", it's completely irrelevant.
 
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Some would ask how do you know -- did you do a carefully setup DBLT between un mod and a modified unit? What are the T&M data to also back those subjective claims? ETC. I get that all the time... mostly from SY. What do you say to that argument?


THx-RNMarsh
Of course not, :). I take a long view over what I read, glean from other's experiences, and combine that with personal exposure - I look for trends in the thinking, and reports of everyone in the game - apply a "filter" as needed, ;), and build up a knowledge base of where things are at. This is no more silly then just grabbing one system, one or two devices to test, then coming up with one set of results - and saying we have now "proven" something ... :cool:.

Edit: The member with the DEQX says that digital in from the CD transport sounds "awful" - which doesn't make normal sense, because there are 2 more conversions happening in the chain. I haven't heard it, so can't comment - but IME this can happen, in the "strange world" of digital audio, :p.
 
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Richard, i don't know about DEQX, but i can confirm the SMPS of my DCX2496 was a pure junk. Noisy with a lot of switching frequency leakeage.
And, as many report they last something like one year before to burn... anyway, better to change-it ;-)

I know..... just giving him a hard time, like what i get. I have a pile of such gear -- just ran out and took this picture --

DSC01673.JPG

I guess most people dont know I have been at this a very long time and have gone thru the iterations, test and measurements that have helped establish many standard mods.... grounding, PS Zo, parts quality, dc servo, CFA circuits, ac EMI/RFI filtering/isolation et al. All the things fas42 does. Now I just do it to better circuits and products. Like I say, it just keeps getting better.


THx-RNMarsh
 
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For those not skilled in correlating harmonics and their levels with audible sound -- this is how I describe what I hear:

A single tone harmonic level of -70 applied to many tones gives the effect of thickening the sound and to some extent to reducing inter-transient silence or reduction in crystal clarity. -60dB harmonic would thicken the sound more. Sometimes higher thd levels will seem like reducing the dynamic range much as broad band noise would when listening to music..... but if a harmonic dominates and with many tones simultaneously (music) there is a freq bias component to the 'noise' which can affect a character to the sound.

When I get masters of 24/96, and compare to CD/16b, the 24/96 recording can be recorded further down and away from clipping -- a common problem for recordest. If I need to be at least -20 down to avoid clipping of pop music or -30 down for other types of music, without compression, it brings me closer to the noise floor. With a 90dB dynamic range on 16b that will put me close enough to the noise that it can be heard. But with a 24/96 recording, I have so much dynamic range that i can record at much lower levels and still be far from the noise floor. And, with techniques which move certain 'noise' above the 20KHz audio range, it makes it harder on the audio OPS to handle the HF 'noise' linearly. That can add its audible affect to the music.

Clipping is so common in recording that there actually exists software that will put the peaks back on a flat top/clip waveform. Thats an interesting trick to enhance effectively dynamic range while staying further from the noise floor. Dont know what that does for the distortion caused, though.

THx_RNMarsh

That is the point of 24b and not audiophile sound. One can set recording level low enough to avoid clipping. Then, in the mastering, recording is leveled up so that peaks are just bellow 0db. 16b is then enough for faithfull reproduction.
 
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Thanks George. Rumors of my death were somewhat exaggerated.

I had asked but got no reply.

And may I ask where is Brad?

I hope you are doing well now :xfingers:


for the letter to Stereophile as one that I didn't thereby need to write. Although I don't know its original length, it says much of what needs to be said about IM distortion, and makes some other good observations about nonlinearities and how they tend to work.

What is the date of that Stereophile issue?


Besides my own recording experience, I get a lot of my recording info from the known Master recording engineers themselves and what they have said/written

Plus –maybe- some help from Internet Archives

https://archive.org/details/gd85-09-07.oade.georges.17078.sbeok.shnf

You can search by Band name, recording engineer, equipment used, etc..
Notice the ‘Source’ and “Lineage’ details, not found –and much missed- in commercial releases...

George
 
Quality of power supplies and voltage rails and references seem to be a major element - anything with a SMPS inside it is asking for trouble, when there are DACs, etc, in the same box. I personally noted that an Oppo used purely as a transport degraded the system sound quite badly in a demo, so some things could possibly be looked at ...
.

Of course all the other electronics in the world with high precision DACs, ADCs sensitive analogue seem to work fine with SMPS supplies... Even in headphone shells powering the electronics...
 
Richard, i don't know about DEQX, but i can confirm the SMPS of my DCX2496 was a pure junk. Noisy with a lot of switching frequency leakeage.
And, as many report they last something like one year before to burn... anyway, better to change-it ;-)

Its so frustrating when you hear this, on quite a few jobs I work on we get my layouts checked with the application guys from the relevant chip manufacturer, doesn't cost anything (they would rather see good implementations of their devices than bad). There is also a wealth of data sheets and guides, though often these are not followed, to guide the layout which is critical. So I find it hard to understand why this happens, the power supply is the heart of a unit, if that doesn't work the rest wont. There does seem to be a move these days for more distributed power, with a main supply feeding many on board local supplies, SMPS for the high current demands such as processor cores, I/O supplies, further sensitive circuitry having further power islands' fed from LDO's.
 
Of course all the other electronics in the world with high precision DACs, ADCs sensitive analogue seem to work fine with SMPS supplies... Even in headphone shells powering the electronics...

But hifi is uniquely different, somehow. Apparently, none of that other stuff can be improved by using wooden cones, cable lifters or tuning fuses :(

They have a lot of catching up to do eh :rolleyes:
 
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