Its very simple. Most sound devices support a number of sample rates. Common ones include 16 bit @ 44.1khz, 16-bit @ 48khz, 24-bit @ 96 khz etc. Only one application has exclusive control over the sound hardware at any time. Whatever rate that application opens the soundcard at, is the rate that is used to send data to the card. If the application with exclusive control over the sound hardware is a mixer-type daemon (e.g. dmix, esd, artsd, etc.) then all audio streams are converted by this application and sent to the sound card using the sample rate it opened the hardware with. Does that make sense? -Pete > On 12-06-08 06:30, Sergei Steshenko wrote: > > >> Yes again - to me ALSA's sample rate implementation looks quite >> illogical - IMO it should be the other way round - user first >> mandates sample rate, and then playback sources adapt through >> resampling if necessary. >> > > Great setup once we have infinitely fast computers to do the resampling > at any quality level we feel like in realtime. Meanwhile over here in > the real world we'll continue on our imperfect attempts to make things > work though. > > Rene. > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > Alsa-user mailing list > Alsa-user@xxxxxxxxxxxxxxxxxxxxx > https://lists.sourceforge.net/lists/listinfo/alsa-user > ------------------------------------------------------------------------- Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
