Are high sample rates better for DSP? Options

[Ethan Winer]

I've read many times that audio plug-ins such as equalizers work better at higher sample rates, even if audio generally doesn't benefit. It seems unlikely that there's a practical advantage to up-sampling all your files, or recording at 96 KHz in the first place, but is there a potential theoretical improvement? Or is this just another Internet myth?

--Ethan

[greynol]

I know the unofficial mantra around here is all things sound the same unless proven otherwise, but all I can say is wow, just wow. When I was an engineer we always joked about something working on an infinite sample of one.

You've tested a couple of DSP operations on extremely trivial single-tone samples and are now willing to make such a sweeping generalization?!? I hope you don't plan on presenting this in public as things might get more than a little embarrassing for you.

This post has been edited by greynol: Jun 12 2012, 18:48

[Ethan Winer]

You've tested a couple of DSP operations on extremely trivial single-tone samples and are now willing to make such a sweeping generalization?!?

No plans to present this in public. I posted this question because a friend asked me about it and I wasn't sure, so I figured I'd ask the experts. Me, I'll be satisfied to hear a good explanation of why EQ works better on 96 KHz audio files than files at 44.1 KHz. I'll be even happier to see an FFT or other data, or even a pair of audio files to compare. Whaddya got?

--Ethan

[saratoga]

Me, I'll be satisfied to hear a good explanation of why EQ works better on 96 KHz audio files than files at 44.1 KHz.

Resampling is accomplished by applying a filter. If you resample to a higher sampling rate, then apply an EQ, effectively the length of the filter in the EQ is expanded by the length of the filter in the resampler. Thus, you're essentially using a filter with more taps, which if properly designed (e.g. a good resampler), will give 'better' (or at least different) results then a filter with fewer taps.

Of course if the EQ is already properly designed for its application, changing it is unlikely to improve things.

[Ethan Winer]

Thanks Drew for some hard evidence. Can you explain in layman's terms what the difference is between the two files after applying the same EQ? From the partial graph it looks like there's more boost at 10 KHz, but that doesn't seem right.

Of course if the EQ is already properly designed for its application, changing it is unlikely to improve things.

I guess this is the crux of it. Are many/most EQ plug-ins written "properly" such that there's no practical advantage to using files recorded at higher sample rates? Does the Sonitus that drew (and I) use do things properly?

As you know, my perspective is what matters in practice. For example, in theory 0.0001 percent distortion is "better" than 0.001 percent, but in practice it doesn't make even a tiny audible difference because both are too soft to hear. I've been using EQ plug-ins for years, and I've never noticed any degradation, or change to the audio other than the EQ I applied.

--Ethan

[drewfx]

Thanks Drew for some hard evidence. Can you explain in layman's terms what the difference is between the two files after applying the same EQ? From the partial graph it looks like there's more boost at 10 KHz, but that doesn't seem right.

The 192kHz (orange) was about +6dB on the spectrum analyzer before the EQ was applied. I turned off the pre-EQ snapshots to make things easier to see, but they were essentially straight lines ~6dB apart.

Basically, the 2 curves are quite different at >15kHz, but you can also see the Q at 10kHz is a little different as well. For the part below 6kHz (covered by the EQ) they appear identical (~6dB apart).

So it's just a somewhat different EQ curve being applied in this case.

And you might note that the 48kHz curve in the analyzer arguably looks more like the curve in the EQ GUI anyway.

[andy o]

I'm guessing these are the reasons why A/V receivers and such oversample for DSP?

[Arnold B. Kruege...]

I'm guessing these are the reasons why A/V receivers and such oversample for DSP?

Do they?