FWIW, I took "talking blasphemy" as irony.

For some the following will be obvious, for others it may be informative.

There are so many layers to the digital audio cake that it's very difficult to give a definitive answer, though I certainly agree that 24bit 48k is probably the minimum we should really be working with. I'd always rather downscale than upscale at the final cut.

Where the definitive bit gets particularly tricky is that the data is not the only factor. Another very significant factor is the behaviour of the convertors to and from analogue. The root cause of the differences at this point are due to the way real sound and the digitisation process can interact. Specifically, if one has an audio sound at, say, 40kHz, it's well outside of the range of most people's hearing, but it you sample that sound, whether at 44.1kHz or 48kHz, you will get an interference signal from it right smack in the middle of the audible bandwidth. A similar but usually less serious issue arises at conversion back to analogue.

The analogue to digital conversion must filter out any signals like that before taking the samples. The problem that then arises is that filters themselves also affect the sound and the harder one makes the filters work to stop those unwanted frequencies, the more the effects of the filter itself become audible.

The faster one can sample, the less severe those filters have to be, so a 192kHz sample rate will be cleaner than a 96kHz sample-rate will be cleaner than a 48kHz sample rate. However after that point, downscaling the faster rates to 48kHz should make no real difference to the content.

When comparing 96kHz to 48kHz the question then arises "what causes any difference to which you are listening ... the sample-rate or the filter?" (or indeed the expectation). And unless one can answer that, the comparison is of dubious value.

Any time we manipulate the content of the data stream we degrade the signal a little. Our DAWs probably work 32-bit float, which helps to minimise that, but it would seem a great shame to tie the output from the DAW to 16bit 44.1kHz before sending it for a final mixdown. That's about as low a quality as one would want to go, but the mix guy will manipulate it further.


There are occasional threads about support of FLAC audio files. FLAC would allow 24bit 48kHz data to be compressed to a similar size to 16bit 44.kHz WAV, so media size should not be an issue. Old computers might be, though increasingly less so I would have thought.

Jazz relative beginner, starting at a much older age than was helpful.
AVL:MXE Linux; Windows 11
BIAB2025 Audiophile, a bunch of other software.
Kawai MP6, Ui24R, Focusrite Saffire Pro40 and Scarletts
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