On Fri, Dec 25, 2009 at 4:38 PM, Bearcat M. Sandor <hometheater@xxxxxxxxxxxxxxx> wrote: > sound card: IntelHD > alsa version: 2.6.31-r6 > pulseaudio version: 0.9.19 > mpd version: 0.15.5 > disto: gentoo amd64 > Sound file format: Flac > > Folks, > > What i would like to be able to do is to play music files at their > native rates. I realize that my sound card's rates begin at 48khz, so > 44.1khz files may have to be upsampled in any case. That's not > preferable but that's alright. However, i'd like to be able to play 96 > khz files and 192 khz files at their respective rates with out resulting > to upsampling. If i have to upsample everything to 192khz to avoid > downsampling files to 48 khz i'll do that but it's not ideal. i can't tell if you're confused or not. the hardware is running with a sample rate we can call "HW". you have files at a different sample rate, we'll call it Y. At some point, something is going to have to convert the data stream that is at Y to HW. this has to happen before it hits the hardware. there are lots of different bits of software that can do this. the simplest is to rely on the audio stack to take care of it. the best, as far as quality goes, is to use either sndfile-resample or sox to create a new on-disk representation of the data. if you rely on the stack, then do not bother asking which part of it is doing the conversion, and don't expect to have any control over its quality. --p _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
