High Resolution Audio: Does It Matter?

[j_j]

For all the assembled brain power, this thread is mostly filled with strutting and posturing, and very low on information and education.

Well, if somebody brings in something with some meat, there will be actual discussion. Until it's not more of the SOS, well, there's not much to say.

 

[j_j]

Whatever nonsense we come up with, tongue-in-cheek, has been done. It's like Poe's Corollary of Snake Oil Completion:

Liquid Audio Cables – Teo Audio

I'm disappointed, the cables that had 6 light fibers in the dielectric around the center conductor in order to "illuminate the electrons" or some such (sorry, I forget the exact claim, but it was in that range, something about "ordering the electrons") appear to have disappeared from the market. The claims were, well, "interesting". And then, back in rec.audio.* there were the mercury-filled speaker cables. That ended somewhat badly, as I understand, but not due to the conductive nature of mercury. (cringe)

Perhaps a separate thread somewhere on "get a load of this" is called for?

 

[ahofer]

I'm disappointed, the cables that had 6 light fibers in the dielectric around the center conductor in order to "illuminate the electrons" or some such (sorry, I forget the exact claim, but it was in that range, something about "ordering the electrons") appear to have disappeared from the market. The claims were, well, "interesting". And then, back in rec.audio.* there were the mercury-filled speaker cables. That ended somewhat badly, as I understand, but not due to the conductive nature of mercury. (cringe)

Perhaps a separate thread somewhere on "get a load of this" is called for?

There's the extreme snake oil thread. The cables from Teo are the stuff they use now in thermometers instead of mercury ("a room-temperature molecular slurry of Gallium, Indium and Tin").

 

[j_j]

There's the extreme snake oil thread. The cables from Teo are the stuff they use now in thermometers instead of mercury ("a room-temperature molecular slurry of Gallium, Indium and Tin").

I'm almost afraid to look.

 

[voodooless]

About academic rigor:

 

[j_j]

About academic rigor:

Yes, and many other things. The point, simply put, is that people who this kind of thing GET CAUGHT, and it does not end well for them. So. your point?

 

[voodooless]

Yes, and many other things. The point, simply put, is that people who this kind of thing GET CAUGHT, and it does not end well for them. So. your point?

The point? That you should always look into the details, never accept stuf at face value. I know, this is not what academics tend to do (as in: they do tend to look into the details), but many other people do. It was more a commentary on the now banned commenter:

And your purported reduction to 'face value' devalues the peer review process under which Reiss's and other papers are qualified in the Journal of the Audio Engineering Society and other academic and professional publications.

Surely the fact that they were caught it great. But they were not caught by “the process”, but rather by a few people that found a fishy smell and wanted to investigate. You could argue that that is also part of “the process”, but some of these papers are more than a decade old. That’s way too long and I’d call that a failure of the academic process.

It’s funny btw how the lack or rigor was even prevailing in the way these deceptions were done. It was really lazy and stupid. Even I could have done a better job at hiding these manipulations, and I’m a statistical(or statistics, you choose) moron . Also, I can do copy/paste in Excel

That these things happen is only Human. Despicable specimens can be found everywhere, also in academia.

 

[j_j]

The point? That you should always look into the details, never accept stuf at face value. I know, this is not what academics tend to do (as in: they do tend to look into the details), but many other people do. It was more a commentary on the now banned commenter:


Surely the fact that they were caught it great. But they were not caught by “the process”, but rather by a few people that found a fishy smell and wanted to investigate. You could argue that that is also part of “the process”, but some of these papers are more than a decade old. That’s way too long and I’d call that a failure of the academic process.

It’s funny btw how the lack or rigor was even prevailing in the way these deceptions were done. It was really lazy and stupid. Even I could have done a better job at hiding these manipulations, and I’m a statistical(or statistics, you choose) moron . Also, I can do copy/paste in Excel

That these things happen is only Human. Despicable specimens can be found everywhere, also in academia.

Well, there's another problem, that of misinterpretation of previous results, for instance, the claim that intrachannel phase response does not matter. The original work used 60Hz and 7000Hz. These two frequencies have zero interaction on the cochlea, the conclusion that "phase doesn't matter" is completely correct. If, however, two tones, or 3 tones, that are within one critical bandwidth, are tested, surprise, surprise, for at least some combinations there is obvious difference. Unfortunately we still see crossovers with phase reversal, etc.

Likewise, there was an argument that LOTS Of phase shift (one million degrees) does not cause an audible difference. The phase shift was introduced via a clever allpass circuit. Yes, indeedy, the phase shift was inaudible. The phase shift, however, was very close to 2*pi*f*t, for a 't' representing about 1 million degrees (divide by 57... to get radians, but it doesn't matter) and in other words, was effectively very near a time delay.

Those of you with access to matlab can try this. Easiest way: set up a very very long spectrum buffer. set (considering sampling rate, etc) the line corresponding to 400Hz to '1'. Set the lines corresponding to +-4 Hz of that to .25 each. Do the ifft (of course you have to mirror the spectrum and all that stuff). Listen. Now change one or the other of the .25's to -.25. Repeat. Listen.

Anyone who says this is subtle, well, you didn't listen.

Of course that is a (*&*(load of phase shift per Hz for 4 hz. Granted. But it is concrete proof that intrachannel phase can matter.

It is, however, still bandied about to this day that intrachannel phase shift can be ignored. (as long as it's the same in both stereo channels). Yeah, the beat goes on.

Oh editing to add yes, I understand the phase reversal need in crossovers, but it can be avoided.

 

[krabapple]

About academic rigor:

"FAKE DATA. Academia is BROKEN!" Watch it on Youtube!

Nah.

 

[krabapple]

We were going there, slowly.

I will say, though, his argument tactics and his way of ambiguous phrasing of everything to give himself multiple outs did not speak well with me in terms of sincere debate.

Yes. IIRC BS's white paper and analogous JAES manifesto noted that 24 bits are better than 16 bits in recording/production situations, and that higher sample rates can make some things easier to do without fear of artifacts-- things that were already known (DAC designer Dan Lavry had already put out his white paper arguing that by ~60 kHz SR, the possible issue of audible filtering artifacts was essentially 'solved'). But BS, not coincidentally the force behind Meridian, was pushing for the desirability of 'hi res' consumer delivery formats, because it would sound better, and that was unwarranted. His argument for audibility in that realm was -- I'll be kind here -- thinly sourced.

 

[ahofer]

Slightly OT, I came across this going through the equipment I have to sell:

 

[krabapple]

All that detail but not a copy editor in sight.

 

[mfnickster]

I found this passage from Bob Katz's “Mastering Audio” intriguing:

The Ultimate Listening Test: Is It The Filtering or the Bandwidth?

In December 1996, I performed a listening test, with the collaboration of members of the Pro Audio maillist. The idea was to develop a test that would eliminate all variables except bandwidth, with a constant sample rate, filter design, DAC, and constant jitter. The question we wanted to answer was this: Does high sample rate audio sound better because of increased bandwidth, or because of less-intrusive filtering?

The test we devised was to create a filtering program that takes a 96 kHz recording, and compare the effect on it of two different bandwidth filters. The volunteer design team consisted of Ernst Parth (filter code), Matthew Xavier Mora (shell), Rusty Scott (filter design), and Bob Katz (coordinator and beta tester). We created a digital audio filtering program with two impeccably-designed filters which are mathematically identical, except that one cuts off at 20 kHz and the other at 40 kHz. The filters are double-precision dithered, FIR linear phase, 255-tap, with > 110 dB stopband attenuation, and < 0.1 dB passband ripple.

After the filter program was designed, I took a 96 kHz SR orchestral recording, filtered it and brought it back into a Sonic Solutions DAW for the comparison. I expected to hear radical differences between the 20 kHz and 40 kHz filtered material, but I could not! Next, I compared the 20 kHz filtered against "no filter" (of course, the material has already passed through two steep 48 kHz filters in the A/D/A). Again, I could not hear a difference! The intention was to listen double-blind, but even sighted, 10 additional listeners who took part in the tests (one at a time) heard no difference between the 20 kHz digital filter and no filter. And if no one can hear a difference sighted, why proceed to a blind test?

I tried different types of musical material, including a close-miked recording I made of castanets (which have considerable ultrasonic information), but there was still no audible difference. I then created a test which put 20 kHz filtered material into one channel of my Stax electrostatic headphones, and the time-aligned wide-bandwidth material into the other channel. I was not able to detect any image shift, image widening or narrowing—there was always a perfect mono center at all frequencies in the headphones! This must be a pretty darn good filter!

As a last resort, I went back to the list and asked maillist participant Robert Bristow Johnson to design a special "dirty" filter with 0.5 dB ripple in the passband. Finally, with the dirty filter, I was able to hear a difference...this dirty filter added a boxy quality that resembles the sound of some of the cheaper 44.1k CD players we all know.

This 1996 test seems to show that a "perfect 20 kHz filter" can be designed, but at what cost? Also note that as this test was conducted in the context of a 96 kHz sample rate, the artifacts of two other 48 kHz steep filters already in use may have obscured or masked the effect of the filter under test. Since I conducted my test, several others have tried this filtering program, and most have reached the same conclusion: the filter is inaudible. One maillist participant, Eelco Grimm, a Netherlands-based writer and engineer, performed the test and reported that there were no audible differences using the Sonic Solutions system, yet he and a colleague were able to pick out differences between filtered and non-filtered blind using an Augan workstation. He did not compare the sound of the 20 kHz versus 40 kHz filters, so we are not sure if he's hearing the filter or the bandwidth, but I believe he was hearing the filter, which must not be ideally-designed. I believe the reason he did not hear the differences on the Sonic system is perhaps its jitter was high enough to mask the other differences, which must be very subtle indeed!

Regardless of whether Eelco's group did reliably hear the bandwidth differences, it should be clear by now that the so-called "dramatic" differences people hear between sample rate systems are not likely to be due to bandwidth, but probably to the filter design itself. Ironically, it was necessary to make a high sample rate recording in order to prove that high sample rates may not be necessary.

 

[JP]

But he doesn’t say if/what controls were used for the “dirty” filter test?

 

[danadam]

This 1996 test seems to show that a "perfect 20 kHz filter" can be designed, but at what cost?

I don't know. Do tell, please.

 

[mfnickster]

I don't know. Do tell, please.

Ask Bob.

 

[j_j]

Ask Bob.

In the modern day, it's trivial. And for the test Bob described, you'd be better off with yet another filter, that one starting to cut off at 25kHz and finishing at 32kHz. Shorter, no possible interference in the time domain based on current understandings of cochlear dynamics. Filter created with the following code:
bb=firpm(160,[0 25/48 32/48 1],[1 1 0 0]);

ripple circa 10^-8 dB, stop band cutoff below 160dB.

And you could reconstruct at 64 kHz just dandy.