For distortion and compression, the only possible difference I can think of is aliasing, and yeah you could just oversample your input, then downsample again on the way out. This is really really common in most professional plugins, often with an adjustable amount to balance quality and CPU usage.

I also have some gear that aliases at a volume you can't hear, but when you plug it into an analog distortion pedal, the aliasing in the high frequencies becomes apparent. This would be avoided if it had a higher sample rate so the aliasing was far out of the audible range.

For other sorts of effects like spectral processing, pitch shifting, the extra detail above 22khz really does make a difference, especially if pitching down.

More bits means:

• Less quantisation noise, so your noise floor is a bit lower and therefore, when you're mixing stuff together, you accumulate less noise

• More numerical room to play with, you can scale or shift the volume up and down more without clipping and with less loss of accuracy

With 16-bit CD audio, you can't just convert to 24-bit and suddenly have less noise. You might get more room, though.

As for higher sampling rates (more kHz, if you will), I think Monty mentioned some benefit regarding less sharp aliasing filters (can have a larger transition band from 20–48kHz, say, rather than from 20—22.1kHz), but it's not something I understand the benefit of well.

filters with a sharper cutoff will 'ring' more, smearing sharp changes in the signal out over time. It's a pretty subtle effect when it's all happening above 20 kHz though

I imagine it's because lossy codecs are tuned for human perceptual limitations, and when you process audio, it can "pull" areas of the sound that are otherwise hidden from your perception into perceptual ranges, analogous to how fiddling with the brightness/contrast of a highly compressed JPEG image can accentuate the artifacts.

We are not really talking about lossy codecs here, though, are we?

Ah, my mistake. But perhaps you could consider 24/192 -> 16/48 a lossy codec in this context; the same argument does apply, for the same reasons.

That way it does make sense, editing might require higher fidelity than we can perceive in the final project to avoid artifacts.

Actually, the article is not talking about it, but the topic very much is lossy codecs.

Specifically, if 24/192 AAC is worth it compared to 16/44.1 AAC (and the answer to that is yes, although the answer to 24/192 WAV is no)

But what about lossless 16/44.1KHz audio? It's not a "compressed" version of 24/192Khz. Chances are that your 192KHz music will be low pass filtered anyway, 192KHz is a trick that sound chipset vendors added so that integrators wouldn't have to make high quality analog filters for their ADCs. There's likely nothing above 22.05KHz except shaped noise, production artifacts, and very, very quiet ultrasound noises which have nothing to do with the music.

There's no such thing as lossless 16/44.1 because it's practically impossible to make an artefact-free antialiasing/reconstruction filter pair clocked at 44.1k.

The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering.

Nyquist is fine in theory, and if you've never actually tried to implement a clean filter you'll likely watch the xiph video and think "Well that makes sense."

If actually know something about practical DSP and the constraints and challenges of real filter design you're not going to be quite so easily impressed.

Likewise with higher bit depths. "Common sense" suggests that no one should be able to hear a noise signal at -90dB.

Common sense is dead wrong, because the effects of a single bit of dither are absolutely audible.

And if you can hear the effects of noise added at -90dB, you can certainly hear the effects of quantisation noise artefacts on reverb tails and long decaying notes at -30 to -40dB, added by recording at 16 bits instead of 24 bits.

Whether or not that level of detail is present in a typical pop or classical recording is a different issue. Realistically most music is heavily compressed and limited, so the answer is usually "no."

And not all sources have 24-bits of detail [1]. (Recordings made on the typical digital multitrack machines used in the 80s and 90s certainly don't.)

That doesn't mean that a clean unprocessed recording of music with a wide dynamic range made on no-compromise equipment won't show the difference clearly.

Speaking from experience, it certainly does.

[1] Technically no sources have 24-bits of detail. The best you'll get from a real world converter is around 22-bits.

> you'll likely watch the xiph video and think "Well that makes sense."

What video? This thread is about an article.

> The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering.

I just said that, I think.

Because modern production can involve hundreds or even thousands of digital effects. So 24-bit, or even 32-bit float just because you can, is standard.

As Monty demonstrates, it's a fraudulent waste to try to sell the result as a product to the end listener.

I think it's because processes that affect things at the individual sample level (which distortion/compression often do) have a lot more detail to work with. The author mentions this a bit in the article