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some times ago i was playing with coding, shannon theoremes and other stuff, i have tried without success to compress audio wave, and i have notice that simply oversampling audio material enacnhe a lot compression ratio i only take awav file, oversampled it by 20 tiimes and then compressi it using pkzip or rar. i don0't remember if i also do a CONSTANT PREDICTION, iony know that pkzip and

constant prediction see this page http://flac.sourceforge.net/format.html oversampling.. i maean digitally change the wave file rate form 44khz to 440 khz it make next sample easyer predictable if i'll found my old work i'll tell you how much compression i could achieve

--- Hod McWuff <hod@wuff.dhs.org> wrote: > On Fri, 2002-10-04 at 10:26, Marco "elcabesa" Belli wrote: > > oversampling.. i maean digitally change the wave file rate form > 44khz to 440 > > khz > > > > it make next sample easyer predictable > > OK, IANASPE (signal processing engineer) but it seems to me that if a > simple shift like that can...

...dramatic changes in the signal. If the guitarist plucks a string, or the vocalist starts a new syllable, that should also mark a frame boundary. On Fri, 2002-10-04 at 13:27, Josh Coalson wrote: > --- Hod McWuff <hod@wuff.dhs.org> wrote: > > On Fri, 2002-10-04 at 10:26, Marco "elcabesa" Belli wrote: > > > oversampling.. i maean digitally change the wave file rate form > > 44khz to 440 > > > khz > > > > > > it make next sample easyer predictable > > > > OK, IANASPE (signal processing engineer) but it seems to me that if...

some times ago i was playing with coding, shannon theoremes and other stuff, i have tried without success to compress audio wave, and i have notice that simply oversampling audio material enacnhe a lot compression ratio i only take awav file, oversampled it by 20 tiimes and then compressi it using pkzip or rar. i don0't remember if i also do a CONSTANT PREDICTION, iony know that pkzip