Hello Max. Thanks for commenting. On Tue, Sep 28, 2010 at 12:57 PM, Max Kellermann <m...@duempel.org> wrote: > I understand your problem, but most of the time, it's the other way > round: people have 24 bit audio (the libmad decoder emits 24 bit > audio!), and their sound hardware wants 16 bit. Your change would do > the opposite, it would lose more precision.
I never thought about breaking existing behaviour. I suggested to add one more convert filter. So advanced users could manually control format before and after filter chain. > What we need is better internal logic for conversion. If you need 24 > bit anyway, sure, convert to 24 bit as early as possible. OTOH, > libsamplerate deals with floating point samples only... It's not so > simple! > > We could make a rule: make that two conversion plugins. Never convert > "down" before the other filters, and never convert "up" after the > other filters. This would get the best our of both cases. It's the almost the same as I thought, but I think manual configuration would be more flexible. ------------------------------------------------------------------------------ Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev _______________________________________________ Musicpd-dev-team mailing list Musicpd-dev-team@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/musicpd-dev-team
