25 Dec
2009
25 Dec
'09
6:02 p.m.
On Fri, Dec 25, 2009 at 4:38 PM, Bearcat M. Sandor
<hometheater(a)feline-soul.com> 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