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Removing Veils in the Audiophile Sound Chain


Ralf11

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John, remember, I am that tweaker. And at a way more microlevel than you.

Not that the rougher level is not important ...

Lush^3-e      Lush^2      Blaxius^2.5      Ethernet^3     HDMI^2     XLR^2

XXHighEnd (developer)

Phasure NOS1 24/768 Async USB DAC (manufacturer)

Phasure Mach III Audio PC with Linear PSU (manufacturer)

Orelino & Orelo MKII Speakers (designer/supplier)

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The more black emerges with less smearing. That one is easy enough. Try random synths or distortion guitars.

 

This is the same as "more air around", I think.

Lush^3-e      Lush^2      Blaxius^2.5      Ethernet^3     HDMI^2     XLR^2

XXHighEnd (developer)

Phasure NOS1 24/768 Async USB DAC (manufacturer)

Phasure Mach III Audio PC with Linear PSU (manufacturer)

Orelino & Orelo MKII Speakers (designer/supplier)

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2 hours ago, PeterSt said:

John, remember, I am that tweaker. And at a way more microlevel than you.

Not that the rougher level is not important ...

Hmmm...   Decoding DolbyA material is a very precision problem, and perhaps one of the subtle, complex problems that one can solve... :-).  It would have been done in software before (accurately -- NO other solution is accurate) if it wasn't so frustratingly tedious.  A simple -- pure SW DolbyA emulation could have been done in yr2000 Intel CPUS if they knew what I know.  All patented or direct implementations are either dysfunctional or incomplete.

 

I gotta worry about 0.25dB in the msec by msec timeframe (no fixed timeconstants, thank you!!!), and control the slew of the changes -- not just the fixed values.  This is hairy/complex stuff -- I wish I could just do a fixed time domain gain matrix and be done with it!!!  It isn't just fixed gains -- if it was, it would have been solved years ago.

 

It would seem that *only* a 15dB gain range should be *easy* -- but it doesn't work that way, the gains have to be in a certain place at a certain time - and it is NOT based upon R/C time constants (well, in a way it does -- but the 'R' varies.)  And, there are two of those screwy variable R/C time constants in series -- each one has a seperately calculated attack and release.   THEN, because of the fast gain changes, I added all kinds of gain trajectory (nonlinear of course) modification -- both fixed AND also calculating in the signal waveform along with it (explcitly designed modulation distortion reduction -- not possible in HW.)  Then, both the HF0/HF1 bands have to have precisly matched (but subtly different) gain changes vs signal, or they create a 'rubbing' kind of distortion sound. * The HF1 band is generated from two seperate input bands, but the higher frequency content suppresses the lower frequency content -- it causes a sound similar to distortion if not precisely correct.

 

All of these complex calculations and others -- the gains end up being within about 0.25dB of a target goal based only on an ancient HW design without an accurate design spec (the trajectory modified/optimized in a very controlled&nonlinear way +-- but that optimization can be removed, because if you use the RAW DolbyA trajectory, you get increase in modulation distortion.)  Of course, I did full HW models when it might be helpful -- most programs only give numerical results, so I had to derive the curves myself in some important cases.

 

The only reason why I stopped at 0.25dB is that the recording engineers said that it was good enough -- DolbyA units are only accurate within about 1.0dB or so (they vary from model to model -- I based mine on the cat22 for gain values.)

 

*  All of this precision, and there is NO DolbyA meter ANYWHERE that can be used to tweak it in.

 

Most recordings are not dead-on accurate, especially anything NR through a Dolby *anything* unit.  It IS futile to get perfection with commercial recordings -- unless they are very special (not remastered or typical audiphile/high res junk.)

 

I wish one could just tweak an EQ/phase curve and keep on tweaking to ones heart content.  NOW, I am tortured by the fact that the code is released, working 99.99% perfectly and tweaking the code is now done only with great care.   Gotta start on C4 -- and I dread it.

 

 

John

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1 hour ago, PeterSt said:

Hey John, you should be able to revert MQA !

😉

 

 

It would be nice to 'fix' it -- I think the term 'neuter' would be appropriate :-).  If I could find the algorithm for 'neuter' -- then I think that most correspondents here would be happy :-).  Back when I actually touched hardware -- the hardware equivalent would be a pair of diagonal pliers applied in the correct place.   Then...  remove the cancer,  problem gone :-).

 

John

 

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55 minutes ago, Ralf11 said:

yes, but what causes it?

 

 With Computer audio it is related to how electrically quiet the computer is, which can be from Processor activity when doing conversions, for example from .flac to .aiff/wav , or even RF/EMI radiated from PWM motherboard controlled fans and their leads  as John Swenson has found . 

Even the use of double screened internal SATA cables of the minimum length needed  instead of the original  wires all side by side type draped everywhere may result in make a small but audible improvement . 

 

 Not that the usual suspects will have a bar of this though. They insist that USB Audio is perfect too, without the use of higher quality USB cables, Iso  Regens etc. and any reports to the contrary are purely anecdotal and worthless :P

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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13 hours ago, Ralf11 said:

yes, but what causes it?

I would say that electronically, it’s distortion. Acoustically, it could be grill cloths that aren’t acoustically transparent enough, or speaker systems with radically deficient midrange, or have defective drivers. In source material it can be anything from poor microphone technique, adverse acoustics in the recording venue, incompetent mixing or EQ, to overuse of volume compression or electronic distortion again.

George

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1 hour ago, ASRMichael said:

If you have a room with acoustic panels then remove a little, it will bring back some atmosphere. 

Or add some diffusers?

System (i): Stack Audio Link > 2Qute+MCRU psu; Gyrodec/SME V/Hana SL/EAT E-Glo Petit/Magnum Dynalab FT101A) > PrimaLuna Evo 100 amp > Klipsch RP-600M/REL T5x subs

System (ii): Allo USB Signature > Bel Canto uLink+AQVOX psu > Chord Hugo > APPJ EL34 > Tandy LX5/REL Tzero v3 subs

System (iii) KEF LS50W/KEF R400b subs

 

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