Audio from vinyl (24 / 96-192 khz). Converting/downsampling? Options

[Taishou]

Hi again

I have several digitized vinyls in FLAC, at 24 bit, 96-192 khz. Is it worth all that sample rate? I was thinking of making CD quality (16/44.1) FLACs or even mp3 320. I doubt I'll ever in my life spend more than, idk, 200 usd in headphones, so the over-quality looks a waste. What do you say? I'm far from an audiophile and I feel pretty happy with mp3 320 (I can't even tell from FLAC).

Anyway, I need to re-encode that at least for my mobile devices. Should I have any special care when converting? I use fre:ac (LAME 3.99.5). Should I just set 44.1 khz when encoding there and done? Or should I downsample with another tool and then encode? I just tried encoding something setting 44.1 in LAME and the result sounds the same as the original 24/192 (with my crappy headphones at least).

The same goes for FLAC or ogg (I tend to use the latter for my DAP). Just setting the sample rate when encoding is enough or is it better to do it with another program? (I doubt, but I wouldn't like regretting later because I made a crappy conversion).

Thanks!


(maybe this should go in the "vinyl" forum, but my concern is more with those high sample rates and downsampling, regardless the source)

[Taishou]

Well, I decided to leave it as it is.

I tried the original 192 khz file with Audacity and my resampled non "-G" resulting file. Both have 2 or 3 of peaks that touch the "1" on Audacity's scale. And both sound the same. I picked a file that the peaks weren't because of cracks/clicks, but during some high note. Headphones, speakers, whatever, there's no difference. Plus, the peaks are too small to be noticeable. You have to zoom all in until you see milisecs on the scale, and then the peaks last for nothing.

I also tried the song with the -G command in SoX. It makes those peaks sit right just below -1, making a volume reduction of ~0.03-00.5 db. As always, it all sounds the same, no matter how hard I try. And even if I try, the sound from the vinyl, although it's great considering the source, is far from perfect, so I guess those little clippings will never matter.

It must be because of what pdq says, that the original shows a peak of 0.9999 and the resampled file is 1. For example, one file:

Original: -2.77db, 0.999900
Resampled: -2.80db, 1
Resampled with -G: -2.79, 0.999969

Original and resampled look the same in Audacity, the -G one shows a little lower. It's kinda puzzling, there's a 0.03 db difference between original and resampled. Yet the graphs/curves look exactly the same. When using -G the difference is only 0.01 db, but you can see the graph lower (less volume).

In the end everything is too insignificant, and when you see commercial CDs like this



With peaks around 1.10 and -10 db, you feel nothing seems to matter much (yes, I've been reading about loudness wars)