hi i've written a small openAL/sdl application that plays multiple sound files at once. all the sound files are wav (openAL does only wav and ogg) and thus quite big, so it wanted to make them smaller (thus loosing sound quality but i don't mind too much). the only tool i've got for this is "nero wave editor" which let's me change frequency and bits (all files are mono). the original files have 44k frequency and 16 bits. what values can i bring those two parameters down without loosing too much quality? thanks for help

If your signal is bandlimited (like speech) you can lower the sampling frequency without loss of sound quality.

Encoding the wav as ogg will make them a lot smaller. If you don't apply extreme effects on sounds there will only be tiny perceived differences to the original for 1:10 compression ratios.