audiophile sound on FreeBSD ?
ralf.mardorf at rocketmail.com
Thu Apr 26 05:10:28 UTC 2018
On Wed, 25 Apr 2018 20:04:52 -0400, Waitman Gobble wrote:
>Hardly anything recorded in a studio the past 20 years (30?) is over
>48KHz. The actual master digital recordings are not hi definition. You
>can find modern classical music recordings with higher resolution. But
>there are also great recordings that were originally analog (like up
>until the 1990's) and digitized at 96/192 and you (well at least I)
>can tell the difference.
>(But there are also people who claim there is no difference in
>shooting JPEG format photos compared to shooting RAW, they claim they
>cannot see any difference when it's quite obvious there's more detail
>If you want 96/192 on a pci card you'd need something several years
>old, verify the chipset and use OSS on FreeBSD.
>There are a couple good USB audio devices that do 96/192. I do not
>think you can get 96/192 out of any "Creative" device on FreeBSD.
Any comparison with a lossy compression image file format is utter
nonsense. Apart from working around latency issues for PA systems 96 and
192 KHz gains you nothing. For listening 48 KHz 16 bit already would do
the job and for production 48 KHz 32 bit floating point should be used.
32 bit floating point doesn't provide a better audio quality, it just
makes production easier. For production usually uncompressed audio
formats are used, but usually not raw header-less PCM, I suspect most
common is WAV. However, regarding all the double-blind test
discussions, lossless compressed audio format are lossless and lossy
compressed audio format are lossy. It doesn't matter how many people
are able to hear a difference or if they should hear a difference to
claim that the culprit is a bad algorithm. To avoid any issue for any
person, using whatever algorithm is provided by the software, lossy
audio formats shouldn't be used.
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