[MPlayer-dev-eng] libac3 in mplayer - how much forked?

[Anders Johansson]

Hi,

> On Saturday, 24. November 2001 22:46, you wrote:
> > Hi Felix,
> >
> > > you are writing nonsense again Andre.
> >
> > What I want to know is if I will be able to use the regular audio output
> > device and have it automatically up/downsample the audio for me? I don't
> > know if that is the right term for it, but I want audio that is recorded
> > in lower than 48khz to be played at 48khz (because of this crappy
> > soundcard).  I know SDL can do that, but it segfaults for me and I have not
> > had any reactions to my bugreport.
> this should be possible aith ao_plugin and it's resampler in new ao2 code by 
> Anders, but dunno if it's ready to use already...
> -- 
> Best Regards,
> 	Atmos

I had a look at the SDL code it can do 2 times up or down sampling but
they don't lowpass filter the sound after adding or before removing
samples so it sounds like crap. I tested it cause I have one of those
fix frequency soundcards myself at home. On my machine SDL only hangs
when I exit mplayer :) I have written resampling routine that can do
fractional resampling which I am currently integrating into mplayer.
It's going to take me another 1 to 2 weeks more or so, cause I need to
write some other stuff first to make it work with all sound formats.
If you feel like helping out just send me an email. I could especially
need some help with testing and optimisation.

//Anders