[f5ve]

If you're thinking "The highest rate I need my signal to be able to replicate is X, so I should set my sampling rate to 2X," then you're wrong and this article gives several reasons why.

As far as I can tell, though, it doesn't mention what may be the most important reason (especially to the folks here at hackernews): resampling and processing.

This is why professional grade audio processing operates at a sample rate many multiples higher than human hearing. It's not because of the quality difference between, say, 192 and 96 kHz, but rather if you're resampling or iterating a process dozens of times at those rates, eventually artifacts will form and make their way into the range of human hearing (20 kHz).

[shampto3]

You’re right, but I fear this idea has become prevalent in audiophile communities where they only want to listen to files that are 96kHz or higher.

In my opinion, having a high sample rate only really matters during the production phase and does not have a noticeable effect on the final form factor. If the producer uses high sample rate during the creation process, I see no reason why the listener would care if the file they’re listening to is higher than even 44.1kHz unless they are planning on using it for their own production.

[duped]

People should prefer 48k over 44.1 but not for fidelity. It would just make the world a better place if 44.1k audio files died out. The reasons it was chosen are invalid today and we're stuck with it, and now every audio stack needs to be able to convert between 44.1/88.2 and 48/96 which is a solved problem, but has a tradeoff between fidelity and performance that makes resampling algorithms a critical design feature of those stacks.

All because Sony and Philips wanted 80 minutes of stereo audio on CDs decades ago.

[hunter2_]

The same could be said about bit depth: 24 bits offers far less quantization artifacts than 16 bits, and those artifacts can readily show up during production processes such as dynamic range compression, but they are extremely well hidden by dithering with noise shaping which gets applied during mastering so ultimately listeners are fine either way.

However, any type of subsequent processing in the digital domain, even just a volume change by the listener if it's applied digitally in the 16 bit realm (i.e., without first upscaling to 24 bits), completely destroys the benefit of dithering. For that reason, we might say that additional processing isn't confined to the recording studio and can happen at the end user level.

I'm unsure whether this same logic applies to sampling frequency, but probably? I guess post-mastering processing of amplitude is far more common than time-based changes, but maybe DJs doing beat matching?

[rcxdude]

Not just eventually: many effects, such as basically any non-linear mapping like a distortion, will create overtones that will immediately alias down if you are not oversampling. You either need to use some DSP tricks or oversample (usually a mix of both) to avoid this happening, which often happens in just one step of an effects chain.

[scns]

A great explanation by Dan Worrall:

https://www.youtube.com/watch?v=-jCwIsT0X8M

[markkitti]

Even the term "oversampling" implies that sampling beyond Nyquist rate is excessive. I think you would agree that one is not being excessive. It is necessary to sample well beyond accepted "Nyquist rate" in order to reconstruct the signal.

[jmsgwd]

I think you're mixing up the effects of _sample rate_ and _bit depth_ here!

Everything you said about sample rate applies more to bit depth. Higher bit depth (bits per sample) results in a lower noise floor. When audio is digitally processed or resampled, a small amount of noise ("quantization distortion") is added, which accumulates with further processing. This can be mitigated by working at higher bit depths - which is why professional grade audio processing routinely uses 24 bit formats (for storage) and 32-bit or 64-bit floating point internally (for processing), even if the final delivery format is only 16 bit.

Sample rate, on the other hand, affects bandwidth. A higher sample rate recording will contain higher frequencies. It doesn't have any direct effect on the noise floor or level of distortion introduced by resampling, as I understand. (It could have an indirect effect - for example, if certain hardware or plugins work better at particular sample rates.)

A survey of ~2,000 professional audio engineers done in May 2023 showed that 75% of those working in music use 41.1 kHz or 48 kHz, while 93% of those working in post production use 41.1 kHz or 48 kHz.[1] These are the basic CD-derived and video-derived sample rate standards.

From this it's clear that even in professional audio, higher sample rates are a minority pursuit. Furthermore, the differences are extremely subjective. Some audio engineers swear by higher sample rates, while others say it's a waste of time unless you're recording for bats. It's very rare (and practically, quite difficult) to do proper tests to eliminate confirmation bias.

[1] https://www.production-expert.com/production-expert-1/sample...

EDIT: add link to survey.

[gumby]

Yeah, a lot of people think “Nyquist” is a synonym for 2 and stop thinking further.

[Espressosaurus]

Heh. Then they don't actually understand what it implies.

Which makes sense I suppose.

[flyinghamster]

Also, when the sampling rates get extreme (software-defined radio), it is well worth moving to complex samples. Doing so allows you to use a sampling rate equal to your theoretical maximum bandwidth, instead of 2x. That's not such a big deal at audio bandwidth, but when your Airspy is slinging a 6 MHz chunk of spectrum, it becomes a huge deal.

[charcircuit]

Another factor which I don't see mentioned is that the tech audio signal is not always directly going to your ear. Is it possible for the sound bouncing around the room to break such assumptions?

[throwaway0665]

This is all about representing signals inside a computer. Audio played from a speaker (or as it exists in the physical domain) is continuous and your ear doesn't have a sample rate. So there's no concept of a Nyquist limit or aliasing with physical sound.

[rightbyte]

Higher sampling rate makes it easier to identify non-sound disturbances. Like vibrations or electrical, that can show up in multiple orders of some frequency.