On Thu, Sep 4, 2008 at 12:28, Carl-Erik Kopseng <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]>
_______________________________________________ Linux-audio-dev mailing list [email protected] http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
