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