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duped 6 hours ago

Higher sample rates are lower latency for the same block size and resampling is not "free" (pick 2: performance, aliasing, latency) so there can be advantages to working with audio archived at higher sample rates.

But all the advantages come down to professional or editing use cases. There's next to zero advantage to using it as a storage format for listening. Just like 24 bit audio (do you have an amp with 96dB SNR?).

Just personally, I have seen little evidence (personally, professionally, or academically) that there is any advantage for lossless audio for consumer applications. For professional applications there are plenty, and it's endlessly tiring to convince people that "no, actually I need 96kHz for my use case."

Where the audiophiles have _some_ argument here is the design of reconstruction filters which I've heard alleged can perform better in the audible frequency range if the stop band is outside of it. But I have never personally tested this, nor cared enough to. But the theory is sound.

Whether or not it's perceptible depends on what you're measuring, though. In theory, there should be perceptual differences in sound localization if your DAC's reconstruction filter is at 24kHz vs 48kHz since it will change the group delay in a critical frequency region, where you'll get sound at >~2kHz arriving later at the lower sample rate. I think it would be extremely hard to test this though, because humans are really shitty at sound localization to begin with, and practically speaking most recorded material is processed to shit in that frequency range to intentionally decorrelate the channels for the perception of "width."

amluto 4 hours ago | parent | next [-]

> Higher sample rates are lower latency for the same block size

This a truly bizarre statement. On the one hand, of course higher sampling rates are lower latency for the same block size measured in samples. But all sampling rates have (almost [0]) identical latency for the same block size measured in time and lower sampling rates allow less computation for those shorter blocks.

[0] If you are concerned about needing to know future samples in order to calculate the actual signal amplitude at a time between samples, then (a) this matters less at higher sampling rates and (b) this is at most a small number of samples and we're talking about block sizes that presumably exceed, say, 5, so this isn't really a big deal.

duped an hour ago | parent [-]

The unit of a block size is samples (frames, technically), not seconds. When configuring audio devices for playback you tune both sample rate and block size for latency. It used to be far more common to tune sample rate than block size alone for tracking. This is getting into the weeds of actual devices though.

Also to your point, this is why compliant peak meters use a mandatory 4x upsampling at 48k.

Dylan16807 4 hours ago | parent | prev | next [-]

> Higher sample rates are lower latency for the same block size

And if your goal is latency, it makes far more sense to change the block size rather than the sample rate.

> But all the advantages come down to professional or editing use cases.

That sounds about right.

toast0 4 hours ago | parent | prev [-]

> Just personally, I have seen little evidence (personally, professionally, or academically) that there is any advantage for lossless audio for consumer applications

I think the advantage of lossless audio is for archival: rip once, archive as lossless; then you can reencode your library with the latest and greatest lossy encoders over time, or just use the lossless if your player can manage it, cpu and storage is less of a limiting factor for players than 20 years ago.

I don't know how many people are actually managing their libraries these days though, so I dunno if makes a huge difference.