>> 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! _______________________________________________ Linux-audio-dev mailing list [email protected] http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
