4 Sep
2008
4 Sep
'08
9:51 a.m.
On Thu, Sep 4, 2008 at 12:28, Carl-Erik Kopseng <carlerik(a)gmail.com> wrote:
preserving
You can use Fons' Zita-resampler:
http://www.kokkinizita.net/linuxaudio/zita-resampler/resampler.html
--
Anders Dahnielson
<anders(a)dahnielson.com>
Regarding the downsampling I would like to know if I
would get any
funny artifacts when downsampling 96kHz material to 44.1kHz (not even
division). Would I be better of to convert to 48kHz for 96kHz
material?
FWIW, I would think 48 kHz would be a better approach, as you'd be (marginally) better quality from the original 96
kHz source (not to
mention
having to mess around with padding bits and other
hackery that MPEG uses
to
make 44.1 work).
I read quite a few places (like hydrogenaudio.com) that you generally
get better encodings (less artifacts) by resampling to 44.1 instead of
48khz *when using lame*, because it is optimized for 16bit 44.1khz
encoding of mp3s.
Is libsnd capable of resampling and adjusting the bitwidth from
96khz/24 to 44.1khz/16, or would I, as you said, have "mess around
with padding bits and other hackery"?