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Fascinating article. Is there any reason to believe that upsampling an audio file should produce a better analog signal than playing it at its native sample ra

Fascinating article. Is there any reason to believe that upsampling an audio file should produce a better analog signal than playing it at its native sample rate? I find that playing high quality mp3's through an upsampling DSP filter and a 24/192 DAC seems to produce a better listening experience. (As the article points out, this could be due to confirmation bias, or the filter making the music a tiny bit louder.) Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true?

zippergz 12y ago
Maybe it's just a better DAC overall? Even at the same bitrate, I think a better DAC can make a difference.
- joosters 12y ago
  That's circular reasoning - surely the definition of a better DAC is one that makes a difference? Anyway, the author also makes this point in the video referenced by other commenters. The computer hardware he demonstrates with is mid-range at best and yet the DAC and ADC are far more than adequate.
masklinn 12y ago
> Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true? No. Watch the Digital Media Primer and Digital Show & Tell, Monty (the article's author) explains how digital sound encoding works and why the analog signal will be reproduced beyond your ability to notice imperfections either way: http://xiph.org/video/ http://xiph.org/video/ And that's for native 24/192, in your case since it's been upsampled the 24/192 signal can't have more information than the original 16/44.1, since that's all the information that went into it.
- LnxPrgr3 12y ago
  The engineering reason to upsample is to simplify the analog anti-aliasing filter on the other side of the conversion by giving it a wider transition band to work with. It also means one analog filter can handle a wide range of samplerates. masklinn is right though—from a consumer perspective, there's no real reason to care how the conversion's done. Hopefully your DAC was designed by a team that knows what they're doing and will handle whatever rate you feed it. Of course, if you have one of those obnoxious devices that can't actually clock below 48kHz or a mixing daemon configured to run at 48kHz, you're stuck oversampling anyway, and depending on your environment that might be done anywhere between amazingly well and linear interpolation. Many years ago I was stuck playing that game on Linux—in that case, there's an audible benefit from using a better designed upsampler in your player. But if you're equipment's not broken, it's a waste of CPU cycles.

zippergz 12y ago

Maybe it's just a better DAC overall? Even at the same bitrate, I think a better DAC can make a difference.

joosters 12y ago
That's circular reasoning - surely the definition of a better DAC is one that makes a difference? Anyway, the author also makes this point in the video referenced by other commenters. The computer hardware he demonstrates with is mid-range at best and yet the DAC and ADC are far more than adequate.

masklinn 12y ago

> Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true? No. Watch the Digital Media Primer and Digital Show & Tell, Monty (the article's author) explains how digital sound encoding works and why the analog signal will be reproduced beyond your ability to notice imperfections either way: http://xiph.org/video/ http://xiph.org/video/ And that's for native 24/192, in your case since it's been upsampled the 24/192 signal can't have more information than the original 16/44.1, since that's all the information that went into it.

LnxPrgr3 12y ago
The engineering reason to upsample is to simplify the analog anti-aliasing filter on the other side of the conversion by giving it a wider transition band to work with. It also means one analog filter can handle a wide range of samplerates. masklinn is right though—from a consumer perspective, there's no real reason to care how the conversion's done. Hopefully your DAC was designed by a team that knows what they're doing and will handle whatever rate you feed it. Of course, if you have one of those obnoxious devices that can't actually clock below 48kHz or a mixing daemon configured to run at 48kHz, you're stuck oversampling anyway, and depending on your environment that might be done anywhere between amazingly well and linear interpolation. Many years ago I was stuck playing that game on Linux—in that case, there's an audible benefit from using a better designed upsampler in your player. But if you're equipment's not broken, it's a waste of CPU cycles.