Traditionally Bluetooth audio meant decoding and reencoding it into a crappier codec before transmission. So it's an efficiency and quality win.

I think some Google Pixel Bud Pro earphones do this for Opus but that is rarer (there's a few other codecs that have been done like this over the years by different manufacturers).

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.

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.

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.

Unless you also have a pretty decent monitoring system the group delay of the speakers isn't going to be consistent so the filters before them wouldn't matter all that much...

Even in that case I would have a hard time believing that any human in a blind test would be able to perceive a group delay of even 360deg above 2k...

You are talking about sub milliseconds differces in the time frequency content arrives at the ears, just tiling your head slightly will have a greater impact...

For capturing analog signals, 2.5X is enough headroom.

The 5X recommendation is probably for digital signals where the frequency refers to the baud rate, not the highest frequency coming through. A fast switching digital signal will have components with higher bandwidth than the fundamental. Using a higher multiple of samples (assuming the bandwidth is there) will let you see the shape of the waveform and rise and fall times better.

And even if you could, would the frequencies that all humans lose with age really be all that essential for the enjoyment of music? We are talking about frequencies most instruments won't even produce unless severely abused.

For some reasons in audiophile-land the magic is always in some elusive outer realms and never right there where the important stuff happens. They spend a fortune on speaker cables, while often not giving a second thought on room acoustics beyond the cosmetic. The magic sparkle is all the way in the ultrasonic, while their listening spaces have deep nulls in the mid-range due to comb filtering from reflective surfaces caused by a lack of acoustic treatment.

I love music (enough to have mixed it for a living) and to me it is very clear how the priorities are ordered when it comes to audio fidelity:

1. Room Acoustics

2. Speakers

3. Electronics & Digital

Going from the back: Assuming you don't get the cheapest of the cheapest and don't abuse the gear by making it do things it wasn't build for electronics and digital audio nowadays is transparent. That means, it essentially sounds the same if operated within spec. Even a 0,50 € IC will have distortion figures so staggeringly low it is below human perception and equipment is getting better still. A decent opamp can have distortion figures like 0.005 % THD with a linear frequency response all the way up to radio frequencies. There can be challenges with driving very weird speakers or headphones, but if you hsve the right combination of gear it doesn't have to be expensive to be indistinguishably good in it's audio performance.

This means speakers are way more important thsn the electronics before it. Their distortion numbers are multiple magnitudes higher (in the ball park of 3% THD), their frequency response is inherently problematic (often many dBs up and down even in expensive speakers), they will hsve different beaming characteristics st different frequencies, small speakers lack bass, placement is essential, etc. So getting good speakers is important.

But all of this is dwarfed by the impacts acoustics. The position of the speakers alone makes a huge difference. The impact of an acoustically untreated space is severe: you can get a completely smeared time response with deep nulls of 20dB and more while other frequencies are highly resonant. Even a budget speaker won't have problems of that magnitude.

So get some ok electronics, even more ok speakers, but invest the bulk of the money/time into the setup of the room itself.

Many adiophiles have that priority list reversed. Room acoustics suck. You need to measure a lot, add ugly absorbers in inconvenient places, can't place speakers where they look nice and conserve space, but need to place them where they work well acoustically, there is no ideal solution and everything is a compromise. So buying a gold plated HDMI cable and imagining the improvement appears to be better. Only that you might be doing it in a room where a positional difference of a few centimeters changes the frequency response of the listening position massively.

Your approaches are, ride a comment (as you did, 9 hours, no response, likely didn't read) or lean on a frequent flyer (the only privilege I have is increased ability to [dead] obvious spam not caught by filters - but it gets my emails seen)

I sent them an email in past minute - Good Luck! (YMMV)