You can't hear the difference, because it's so subtle that's not worth mentioning... It really doesn't affect anything important as for the resulting sound. And that's what's really important - under normal, usual listening conditions you don't hear any difference.
I don't trust these kind of comparisons, that are not rigorously controlled. Could you give more details?
QuoteQuoteThis snippet has an only, non-ambiguous, interpretation, given that it doesn't contain any frequencies over 22050 Hz ...It believe it does contain frequencies above 22050 Hz since I did not obtain it by properly downfiltering it (from e.g. 192 KHz). But instead by fiddling with Cool Edit in the 44.1 KHz domain.Doh! we were doing so well! If it's sampled at 44.1kHz, it doesn't contain anything unique above 22.05kHz. Anything above this is a copy of what's below it. And anything below it that you intended to be above it, isn't!
QuoteThis snippet has an only, non-ambiguous, interpretation, given that it doesn't contain any frequencies over 22050 Hz ...It believe it does contain frequencies above 22050 Hz since I did not obtain it by properly downfiltering it (from e.g. 192 KHz). But instead by fiddling with Cool Edit in the 44.1 KHz domain.
This snippet has an only, non-ambiguous, interpretation, given that it doesn't contain any frequencies over 22050 Hz ...
QuoteI compared the Cool Edit upsampling algorithm with several of it´s filtering algorithms. I found: Doing the silence fiddling in the 192 KHz domain and then filtering out anything above 22050 Hz looks much better than doing that fiddling in the 44.1 KHz domain and then upsampling it.There shouldn't be any difference. (...) But you should be able to get a very similar result in CEP with the right filter. Really!
I compared the Cool Edit upsampling algorithm with several of it´s filtering algorithms. I found: Doing the silence fiddling in the 192 KHz domain and then filtering out anything above 22050 Hz looks much better than doing that fiddling in the 44.1 KHz domain and then upsampling it.
P.S. CD "sound" (if there is one) is glassy, and less realistic than 24/96. This is what I heard with professional DCS convertors, comparing A>D>A at 44.1kHz and 96kHz (both 24-bits, so both are actually better than CD) with the original analogue signal. (...) I can quite believe record producers who use this equipment everyday (and who hear live music every day) when they say that, to them, the difference is significant.
But almost all of the faults I hear with CDs at home are almost certainly the fault of bad mastering.
If I could have access to some 24/96 music in wav format, I could set up one of those tests.I would downsample the 24/96 wave to 16/44.1 using "realistic" filtering + resampling to 44.1 KHz in CEP (that should be equivalent to the filtering of a good ADC), and convert it to 16 bit with dither. Then, I would play this wave with one of my cards, using its converters running at 44.1 KHz, and record the result with my other card at 48 KHz 16 bit. Ideally it should be recorded at 24/96, but I just have 1 good non-resampling card at 44.1 KHz that is used for playback, and happens to be the same one that also supports 24/96. My other card is non-resampling just at 48 KHz, but is limited to 16/48, although I think has pretty good quality. I think the the fact that the recording is done at 16/48 instead of 24/96 wouldn't have much influence in the test, specially if no differences are found at last. But this point should be analyzed with detail to see what problems it could have, and improved if found necessary. At first, I think that this 16/48 recording would capture all the "nastieties" (aliasing, smearing, etc) of the 16/44.1 DAC, but would introduce some of the 16/48 ADC, although at higher frequencies.Then, I would upsample to 24/96 this 16/48 record, and people with good 24/96 cards could perform some blind listening tests comparing the original vs. the this other wave that has been downsampled, played, recorded and upsampled.At the PCABX site there are some avalable clips to perform tests of this kind ( http://188.8.131.52/technical/sample_rates/index.htm ). They are interesting and show that AFAIR there is no difference in practice, but the clips available are not very representative of real-word conditions. They are very short on duration (and typical fans of hig-res formats won't consider them very representative), and in fact no real-world DAC has been used in their generation, just software processing.
At the PCABX site there are some avalable clips to perform tests of this kind (...) and in fact no real-world DAC has been used in their generation, just software processing.
And you're right in one thing - I am deeply convinced the best audio engineers were in job between 1954 and 1965 (approx.). They played with mikes, they could perfectly record the space, their recordings have depth... One just can't believe what they could do with the relatively poor equipment. The recordings from that era are just unbelievable, marvellous, monstrous, incredible... I have a lot of them (many on vinyls, but they sound good also from good remastered CD) and they make me always joy when I listen to them!
The best procedure for such test seems to be to use one and the same output equipment (soundcard) with one and the same output sampling frequency (96 or 192 KHz, 20-24 bit). And to do anything else only in the digital domain.
Otherwise there will always be analog components that may interfere and that may distort the result of such listening test.
@2Bdecided -- You mentioned in your post that you discussed Demo2 with Halverson, and that he agreed with the impressions you had of a glassy sound with 44.1 and 48 kHz, and of more realism and depth with higher sampling rates.Did he offer any possible explanations for the effect, gear-based or otherwise? It was his equipment, after all, and he must have given it some thought; maybe even played with different gear to see if the effect was consistently reproducible?
David (2Bdecided):I find your report interesting, I hadn't read about it before. But, as you may imagine, I see some problems with it.
First, the program material is not what I would consider "critical". I doubt that a 1960 analog recording will have much content over 20 KHz, if any. And this counts too for the differences you also perceived with the 48 KHz sampling rate, that allow up to nearly 24 KHz to be captured. Still, I could be wrong.
Then, what you talk is interesting, but is not more than anecdotal evidence. Who knows if you could have passed a double-blind, statistically sound listening test, even at the exact conditions you heard the differences.
Even if you passed it, it should be analyzed where the objective differences lie. We should analize if the differences are due to the particular setup you listened to, or just due to sampling rates. First, we could analize if there was some problem at the AD/DA chain. If not, we should analyze (measure) if the differences could be due to some "problems" with the converters at the show (I'm being bad here), or due to DCS converter implementation, or due to common converter implementation. For this, we could also try the high quality downsample/upsample process and compare.
As for why - I can't remember who said it, but most of the ideas that were mentioned are now discussed on the dCS website, or in papers written by my old tutor (Malcolm Hawksford - search for his AES conference papers if you can). Energy dispersion. Non linearities in the equipment and even in the air. The Japanese (?) paper showing the change of blood flow in the brain when ultra-sonic sounds were present was shown first at that AES (without a translation!), and none of us could follow it! There was a general feeling that, well, maybe it's something to do with something we don't know about human hearing (it is so non-linear), but it's much more likely that it's some engineering issue which is explanable with real science if only you track down all factors.
Whilst that is all interesting and vital for scientific research, it's irrelevant to knowing whether 96k is better or not. Let's assume I could pass a blind test. Let's assume some of the other people who claim to hear a difference (they're not all using dCS convertors!) also pass a blind test. "Passing the test" means detecting 44.1k vs analogue, but failing to detect 96k vs analogue. This actually clears up most of your points.
Then, and taking into account all possibilities, the issue would be which perception is more accurate to the real-world experience: the one caused from the signal lowpassed around 20 KHz, or the one that is lowpassed?
Now, consider a non ideal amplifier, where the input/output characteristic isn't quite a straight line. Maybe there's a small kink in the graph around y=x=0. (i.e. the middle, the origin). In a normal system, very quiet signals are always going to hit this kink, because it happens around very small voltages. However, add a huge amount of ultrasonic noise, and any quiet audible signal can be pushed onto any part of the input/output curve, and to different parts moment by moment.
Also, I think the paper lacks some details over the experiment, such as some detailed measurements of the signals at the listening location...
Totally totally agree! Some of the early stereo recordings are so good, it makes you think "how did they do that?!" and why can't we do as well today!Now, you could say that it was the songs themselves that made a difference - true, they certainly did. But there was a magic in the actual recording from the 1960s that was completely absent from the others. I'm sure it was the primative technology, and the comparative simplicity of the mixing process that made the recording distinctive, good, and enjoyable.
There are spectrums measured from the listening location : http://jn.physiology.org/cgi/content/full/83/6/3548/F1