Hey, I thought about extending MP3FS, a user level file system that shows flac files as mp3s to user-space programs (see mp3fs.sf.net), to make it work with my 96kHz/24 bit music rips. MP3FS uses liblame and libflac on the inside, but only converts standard 44.1kHz/16bit files at the moment. Looking at the not-to-well-documented lame library I (think) that Lame only support sample rates up to 48kHz, so I would need to convert the samplerate and bitrate through the use of an other library. I finally found libsndfile today, but thought I might hear with the expertice (that's you), if it should work (if it supports what I want to do),or if I should use something else. I would use libsndflac to convert 24 bit/96kHz and 16 bit/44.1kHz flac files to 16 bit/44.1 kHz uncompressed audio, and then use liblame to convert this to mp3. 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? I do not know much about lame, mp3 encoding, or audio development apart from the basics, so guide me into safer waters if I have drifted into unknown waters here ;) br Carl-Erik Kopseng