Re: [LAD] Coding 96kHz 24bit flac material to 16bit/44.1 mp3

4 Sep 2008

...
   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 preserving
 (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"?

br
carl-erik

p.s. does anyone know why Gmail insists on responding to the person of
the last post, and not the mailing list? I almost replied to fred and
not linux-audio-dev!

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Re: [LAD] Coding 96kHz 24bit flac material to 16bit/44.1 mp3