nipsen

joined 1 year ago
[–] nipsen@alien.top 1 points 11 months ago

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.

[–] nipsen@alien.top 1 points 11 months ago

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.

[–] nipsen@alien.top 1 points 11 months ago

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

[–] nipsen@alien.top 1 points 11 months ago

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.

[–] nipsen@alien.top 1 points 1 year ago

Above a very low threshold level, the sound quality(type reproduction, lack of distortion, actually reproducing the source, etc.) is pretty much the same, yes.

But what you do get is either a) a dac set up in such a way that the expensive components (also on the dac, with soldering and chipset pieces) don't really matter (wiring, plugs, etc), or b) tuning and noise-filter passes that insert a signature to the sound that you can absolutely hear (this also happens in mixing targets, which makes this difficult).

In either case, you can hear a difference for sure. But it is not typically the case that you can discern which one is "quality" and which one is just different tuning. I.e., you might like one over another, and that difference might be worth something to you. But writing it up as a quality-difference is not often a good idea. This has been the case for a while now.

And.. then you arrive at a point where what would cost a fortune 20 years ago (in a very expensive amplifier) can be found in a usb-powered 2x2cm block - and this low threshold level you need to get past to not really be able to hear the difference in quality has truly become extremely low. To the point where expensive dacs might actually be of the a) kind, introducing chipset pieces and soldering that might not be that great. Or the b) kind that tweaks the output - and since the chi-fi option often just lacks the amounts of components and layers to have those mechanical issues, and also lacks "signature" styles and filters, they just are.. dare I say it, objectively better. Because I prefer a less filtered setup if possible (if that doesn't cause issues - which it might). And I know many others (including people I've tricked to like something they would normally have burned with fire and spat on) who also prefer this over massively more expensive setups.

So I think that in that world, even dacs that are 16bit/44,1khz, don't really expose themselves as insufficient (given that that is the source and the target format). But dacs still do sound different, for absolutely certain. And if you target higher bitrate formats, then there are differences in the strategy used to do that that might be interesting depending on what you want to put that signal to.

For me, I think the moment I realized this was maybe a bit of a problem, was when I was led around to listen to a Hegel setup once. I wasn't going to buy something like that, being a sound-peasant - but the shop was empty, and I led the guy on a bit on a journey through sound-technical things and formats and so on. And we ended up trying this setup on different sources, on different inputs, and so on. CD, digital input box from Hegel's stuff. And some random other box, an hdmi source, and a few more sources around the store. I had different lossless and compressed formats to try(from higher sampled sources), and they had demonstration tapes and their own things, so this was interesting to both of us. And I offered as a joke to try using my phone (which had a - I realized eventually, a nice in-built dac) through usb and jack. And we tried that as well.

And I realized that all of these sources basically gave the same sound output. I could put the bitrate and the sample rate higher and lower, and I just couldn't hear it on that setup. I could notice the analog sources were different from the hdmi-source and the other streaming source, but it wasn't because of the sound quality change, but from different filtering and tuning (on that stage, on their dac). Some higher bitrate sources (which were mixed to a higher bitrate target) genuinely didn't come out (except as the classic softening you experience in cheap hifi-setups when the density of the source is too high, or the peaks are moved slightly - that I recognise now as filters and tuning - that now messed with the balance of the mix, highlighting weird stuff that shouldn't be). Sure, the Pink Floyd tracks were good from the high quality stream, but they were also fine on the audio jack on my phone(..presumably now bypassing a comically expensive Hegel Dac and using my crappy phone dac instead, and really adding terrible noise from a very cheap jack.. right..?), including when I turned down the bitrate of the source. Wtf. I didn't believe it, so I had to double-check - is there a problem with the output, is the amplifier involved, is the bitstreaming not working, etc. Surely there had to be some difference, right? It's this brand! It costs a fortune, it has to be magically good!

But that's just how that sound-system was set up, with a couple of stages of careful conversion for the analog inputs intended for noisy sources, and a different layer of filters for the hdmi and other digital inputs. That then was in the end reproduced out at about the same signature, presumably tuned in some way that they would like the speakers to be favoured by (they were not). In 1990, this would have given you a very solid output on a bunch of different sources, of course. And the tuning also clearly favoured the demonstration samples that were higher bitrate formats (but not sampled at higher than their dac.. >_<). All the pieces here were competent hardware as well. But because of the way it was designed and put together at the inputs, it basically cut off anything past cd-quality - even if you played from a higher bitrate source. I.e., if you put a random 16bit/44,1Khz dac on the input to this(or something sufficiently high quality enough to not cause distortion, basically.. like.. no standard whatsoever), it really wouldn't make any difference to what came out, basically regardless of the source quality. It'd be pretty much the same, quality-wise.

And this is a surround-system that cost more than a new car, right? The difference, in the end, between slightly muted and slightly harsher kitchen-player sound: many, many, many moneys.

It just completely changed the way I thought about hifi-sound in general. In the sense that if you are playing back a digitally sampled track mixed to a relatively low bitrate format, there is no /reproduction quality/ difference between something for infinite amounts of moneys and something for very few moneys. Where the difference comes in is when you are able to effortlessly hear all the details in the music at a low volume on a setup - you can pay a great deal before you achieve that (but it's also not necessary if you have a digital source, converted properly, and then amplified to the target speakers competently well). And to then play very loud without inserting artefacts or uncomfortable parts of the spectrums and so on, that is also something you might conceivably pay a lot of money to achieve.

But most people pay for that "signature" that literally has nothing whatsoever to do with the quality of the sound reproduction. Worse, a lot of what is supposed to be really high quality stuff doesn't necessarily favour higher sample sources anyway.

Meanwhile - what is interesting is the maths behind a dac that adjusts towards the resistance of the target automatically (as a pre-amp - ease of use, no need to be a sound-engineer to get things to work). And using several layers and then splice the waveforms later, to be able to convert at a very low effect without creating noise. That sort of thing is interesting, specially for mobile and portable targets (or in a sense also professional mixing stages, because "good enough" is ok to get if you can get it instantly).

But beyond that? No. No difference.

[–] nipsen@alien.top 1 points 1 year ago

You absolutely should try them before you buy. Unless your local hifi-dealer is a bit of a pusher, and uses phrases like "delightful sound", and "this step up in price is worth it in audio quality". Then maybe have them shipped with a return-option. And then test them out (preferably with different sources), without any pressure.

I'm not entirely sure it's worth it to pick a "relatively good+" IEM now, though. Or, if you buy something that is made for a phone(or that should also sound good on a small amp) anyway, odds are that you could get a better and lighter design, that would sound pretty much the same, and cost less than half. That's really where the market has changed - tons of really good IEMs exist, but they're generally all made for that same segment. So perhaps you should consider spending some of that extra money on a custom IEM with an ear-mold piece, or something like that instead.

Unless you have a nice dac and a good playback source, and you go .. by now.. and shop for discontinued higher Ohm IEMs, in what is still a very overpriced segment. There are some nice sets out there. But to be completely honest - I don't think it's worth it. Get those cheaper earbuds that actually fit and are comfy to wear, and then get a good headset or something instead.