>> renice -20 -p `pgrep mpd` >> >> but my Athlon 2.2Ghz still can't handle it for more than a few >> seconds. I don't have SMP enabled because of a bug in madwifi, and >> I'm hoping when I get that fixed I'll be able to run the best >> libsamplerate resampler. Any other ideas for making this work? > > AFAIK resampling is expensive operation that's only necessary when your > sound card can't handle native stream sample rate, furthermore, it's a > lossy operation (degrading quality). > > So, I'd look for the answer to the question "why mpd is doing it and > why I allow it to do that?". > For example, you might have enabled it to resample stream to 32 bits > depth, while your built-in card can only handle 16 and the stream has > also 16, so what happens is userspace-level conversion (with some loss > of quality) to 32, loading your CPU, then this stream goes to alsa, > and, provided that your card can't play this, driver or the card itself > converts it back to 16. > Note that the latter case would probably mean "card offloads conversion > to your CPU as well", so you'll get CPU load for both ways' conversion > anyway, only reducing sound quality, no matter how good converters are. > > To avoid any processing, try disabling resampling in mpd, since it'll > probably be done for you anyway, if necessary (you'll hear "white > noise" otherwise). > > And you can pre-convert all the streams to any given samplerate, but > note that you'll probably get far worse results if the target format > isn't lossless (flac, ape), even if the source one is lossy, than with > worst resampling. > And you can get worse CPU/IO load with lossless format in the end, > since it's harder to decode and the input data stream is much heavier > than with lossy mp3s or oggs. > > -- > Mike Kazantsev // fraggod.net
I'm upsampling my 16/44.1 files to 24/96 because it sounds much better than letting the USB DAC do it. This was actually recommended by the manufacturer and it sounds much better. Pre-converting sounds interesting. I could convert all of my 16/44.1 files to 24/96 files? That way the CPU wouldn't be stressed at playback time. How can I do that? I use libsamplerate "Best" for resampling. - Grant
