On Sun, Dec 23, 2012 at 4:35 PM, Fons Adriaensen <fons@xxxxxxxxxxxxxx> 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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