linuxaudio.org

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

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