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

On Thu, Sep 4, 2008 at 12:28, Carl-Erik Kopseng <carlerik@email-addr-hidden> wrote:

> >> 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"?
>

You can use Fons' Zita-resampler:

http://www.kokkinizita.net/linuxaudio/zita-resampler/resampler.html

-- 
Anders Dahnielson
<anders@email-addr-hidden>

_______________________________________________
Linux-audio-dev mailing list
Linux-audio-dev@email-addr-hidden
http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Received on Thu Sep 4 16:15:02 2008