Imo, in general, the biggest lift in quality you’ll ever get will come from avoiding an analog link between the source and the amplifier, and not having the noise-filters intended to handle that applied to the output. I’m not sure how the Goxlr works, but think it’s a mixing thing for multiple inputs? So it probably has some filters, a range limit, and a pre-amp that will shear off the edges a bit, and put a generally clean sound out. That probably also applies if you’re running usb input for the playback as well.

If you compare that to having some usb-c put to a pre-amp or a dac that has minimal range enough to drive the speakers (200Ohm?). And that also does nothing to the output other than convert it. Or, you have an hdmi-input to an amplifier that just amplifies. Then you’ll hear a difference very quickly.

It won’t be huge. But having a pass-through analog cable to an amp really requires you to add a few filters and tune the input to not get kind of crappy sound, regardless of setup. A mixing board needs to do certain things to the signal on the way. While just tuning the output at the end of a converted digital signal is sort of impossible to get wrong.

So that’s - I think at least partially - why this stuff is somewhat controversial. Because it is the case that with tuning options and mixing possibilities, you can make a lot of things sound significantly less bad if you mix it well. Like one of the sound-tuning wizards I knew that mixed for the local bands - he always managed to do… something, either live or on the recordings that would salvage what was absolute crap to begin with. And it was really impressive. But he would never claim or say that anything of what he did was somehow getting the best sound that could be reproduced. He made that point a few times, that he was pulling filters to bring out certain things to produce a particular sound-picture that would work on the speakers, not that he was somehow drawing on the knobs and getting some magical state where all the waves mathematically align and becomes perfection.

So being able to remove that part of the equation with the analog inputs and the balancing - and preferably also the noise-filters that are in there to avoid amplifying the noise from mixed inputs - is kind of where it’s at, imo.

what are the best in terms of BT headphones?

The ones that are made with the power supply limits in mind, and aim for reproducing AAC or aptx type of streams. Trying to increase the sample rates, upping some number or other, getting higher peak effect – it’s like putting a turbo on a 1l Polo or something like that. Sure, it can get you more power in certain registers. But unless you change every part of the equation, including the usage scenario and the fuel - that car is not actually going to output that power in any useful part of the register for practically all of the time you’re using it. Where you’re also going to just have more noise and whine. Maybe it’s good noise, but it’s not getting you more power on the road.

So very svelte designs. Simple, 16ohm drivers, no extra tuning for “boost” or whatever, is going to get you the best overall results.

Most of the time we’re really talking about the same master mixing target ending up in different formats, though. And although that’s usually different enough to hear, it’s not necessarily just better to hear the difference in quality of some of the sample tracks… or the quality of the noise, the filters on the microphone(and weaknesses of the microphone used), the scratching in the chair, hairs on the bow, flicking of valves, super high definition scratching on strings, and things like that.

If people actually do dig up original mastering tapes to resample them to higher resolution than was done earlier - then that’s great, though. And if a composer or a producer ends up deliberately targeting loss-less with both their samples and then the final mixing target as well, then all of those things would of course add something here that you could potentially enjoy and listen to – that really would be different and better.

But I don’t think that actually is what happens even some of the time. :p

The original Atom Amp+ uses a single LME49600 buffer per channel. Adding parallel buffers increases power proportionally. However, LME49600s are large chips–not exactly optimal. Replacing the single, large buffer with numerous, less powerful buffers achieves the same goal in less space. Parallel buffers also happen to cancel out some common mode noise, pushing the noise floor even lower.

Damn. Take my money now.