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I'm in the process of designing a soundcard. However, I'm a little confused over one aspect. The primary usage of the soundcard's output will be playing 320kbs mp3s through a program (Traktor, VDJ, Serato...). I'm under the assumption that one cannot polish a turd and am struggling to see the point in supporting sample rates of 96Khz and bit depths of up to 24 bits. Any insight?

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The point is to have more resolution than the typical final output so that there is less rounding error. 96khz is chosen because it is exactly twice 48khz which is the standard audio sampling rate for video. This way, you can cut the audio from 96khz to 48khz by cutting the number of samples in half, so there aren't aliasing problems. Similarly, 24 bit audio gives 8 bits more precision than 16. It's the minimum increase that makes sense since you want the samples to be a set number of bytes.

This allows for 256 times the resolution in terms of intensity and twice the temporal resolution while editing and limits rounding error so that the final output is higher quality when it is mixed down to 48khz or 44.1 khz and 16 bit.

If the soundcard is only going to be playing 16 bit, 48khz sources, then there isn't any gain by supporting the additional bits and samples as the device on the other end could accomplish the same by simply sampling the audio at the desired rate and bit depth since the information is already lost, but it makes a lot of sense when using higher quality inputs.

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    "cutting the number of samples in half, so there aren't aliasing problems": This only works if the 96 kHz material is strictly bandlimited to 24 kHz already, in which case the 96 kHz sample rate is useless to begin with. - "24 bit audio gives 8 bits more precision than 16. It's the minimum increase that makes sense": It is also the maximum increase that makes (somewhat) sense, because no speaker will be able to fully reproduce even that much of dynamic range. -
    – chirlu
    Aug 27, 2013 at 11:47
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    "twice the temporal resolution": Increasing the sample rate does not in any way change the temporal resolution of the signal, only allow higher frequencies (from 24 kHz upward, inaudible for humans). - Again, see 24/192 Music Downloads ... and why they make no sense.
    – chirlu
    Aug 27, 2013 at 11:48
  • @chirlu - that post talks about final listening, not processing. Yes, it may not make sense to use it for final output, but it DOES make sense during signal processing. There is a fundamental difference between what machines can do and what we can hear and rounding error does make a difference. This is the same reason that photos and video work with levels of sampling outside what we can see until final output as well. It guards against further quality loss from alterations.
    – AJ Henderson ♦
    Aug 27, 2013 at 13:48
  • @chirlu - as for the temporal resolution, again, this is talking about for processing. See the article you linked in the section on Over-sampling and it's application to filters. Just because it isn't needed for human hearing, doesn't mean it isn't helpful in processing.
    – AJ Henderson ♦
    Aug 27, 2013 at 13:55
  • "that post talks about final listening, not processing. Yes, it may not make sense to use it for final output, but it DOES make sense during signal processing." While this is true, the OP's question is about "soundcard output", making it more concerned about final listening rather than processing...
    – landroni
    Nov 8, 2015 at 13:49
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Not sure about your output, but here's an example of why I would want a sound card to meet or exceed 96KHz at 24 bits on the input side. I have a Zoom G2.1u effects processor and a Tascam DR-08 that both can deliver 96KHz 24 bit audio, so I would want to be sure my sound card could deliver this fidelity to my DAW or NLE.

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  • Hmm that would make some sense in reverse. But in terms of fidelity, what is say 16bit 48Khz going to sound like at 110+DBa
    – Cameron Miller
    Aug 26, 2013 at 23:11
  • Sorry I don't understand.
    – filzilla
    Aug 26, 2013 at 23:22

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