[LAU] Pro Audio? OT rant.

[Thomas Vecchione]

On Sun, Dec 23, 2012 at 4:35 PM, Fons Adriaensen <fons at linuxaudio.org>wrote:

> You seem to think that resampling is some very complex operation
> that involves compromises and deliberate loss of quality, just as
> e.g. mp3 or ogg encoding. That is not the case. As I wrote, it's
> just a filter. It's a bit more difficult to write than e.g simple
> EQ because the coefficients are changing for each sample. But in
> the end the DSP operation that is done is exactly the same as for
> a simple FIR filter, and any loss of quality will be similar as
> well.
>

If this is the case, why would we have several different resampling
algorithms with varying levels of quality?


>
> Which in practice means it's *irrelevant*. Your 24 bit DACs may
> provide 145 dB S/N in theory, in practice even the most expensive
> ones don't even have 120 dB. The limit is the analog electronics,
> and it is not imposed by technology but by physics. Even with the
> best 24-bit converters you can buy at least the lower 4 bits will
> be pure noise. And for a 32-bit converter (snake oil if there ever
> was any), the lower 12 bits will be noise. That's assuming you have
> a high level line input. For mic signals the actual S/N ratio will
> even be lower, much lower in most cases.
>

All very true, and I certainly don't dispute this, but also I am not sure
as to why exactly it is relevant to this conversation either.

      Seablade
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