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
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 ;)
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