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Comment by lz400 | Hacker News Reader

There's a very straightforward practical reason why you might want to keep downloading 24/192. The people who put those together are on average "audiophile" snobs so the rips tend to be perfect, often from high quality sources like SACDs, hdtracks, etc. The source of the recording is usually the best master known for that record (special remastered editions, etc.). If you download mp3/spotify chances are the copy you download is from a worse source. Sure, you don't need 24/192, a well ripped 320kbps mp3 would be the same in practice, but looking for 24/192 in the internet is an easy way of getting better quality music on average in practice.

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> "If you download mp3/spotify"

You are misconstruing Monty's argument here. He is very much against mp3...in fact he says he could tell the difference between high bitrate mp3 and 16 bit 44k wav. The real point of the video is that 16 bit 44k wav is beyond sufficient...don't need to go beyond that to 24-bit 192kHz.

As far as I know, people consistently fail to tell the difference in blind A/B tests in listening between (any decent) FLAC and a well ripped mp3 from the same source. I know I can't. I can't link to proper double blind studies but that's the general consensus in the non-BS-audiophile community as far as I know.

c't magazine did a test all the way back in 2000 (see https://www.heise.de/ct/artikel/Kreuzverhoertest-287592.html) and even the best test listeners were unable to tell 256kbit mp3 and the original apart.

Was this the Arny Krueger stuff? I did that.

I ABXed 320kbit mp3 from an uncompressed original, I think 9/10 IIRC. It was a recording of castanets, and listening for frequency response differences was useless so I keyed off of 'personality' differences in the sounds and did it that way.

I was also just as horrible at detecting wow and flutter, as I was good at detecting lossy compression 'changes of sonic personality'. Comes from my experience being with analog audio, which is less true of people these days.

The idea that 'the best' listeners cannot tell 256k from even something as limited as 16/44.1 is ridiculous. Any mp3 format… any lossy format… is seriously compromised. Talk to techno/house DJs and producers about how useful it is to perform off mp3s, this is not a hypothetical argument.

My anecdote is that I've done a number of these tests (Foobar has a plugin that allows you to do them on yourself), and I can reliably tell the difference between FLAC and 128 MP3, but can't tell the difference on 256 MP3 and up.

Yes, I've heard people putting the threshold in ~192kbps. Above that it's pretty much impossible these days. I can also tell the difference in 128kpbs, although sometimes it's a bit of nitpicking (can only hear differences in the sounds of hihats and things like that)

I think the type of music you are listening to assists with being able to differentiate. To me (much rock, blues, guitar music) 128kbps MP3 sounds like somebody chewed it first. Cymbals and bizarre snares that sound like the snare is now made of paper indicate the low bitrate.

But 256+ and I certainly cannot tell the difference reliably.

The same in electronic music since there are many repeating sounds. 128kbps just sounds bad, especially on a proper sound system.

Any serious club system can produce peaks waaaaay hotter than 120db, cleanly. Especially horn loaded systems like most anything on the high end these days. Horn loaded PA speakers can give you perfectly clear peaks at absurdly high amplitudes, cleanly. We don't hear peaks, we hear RMS volumes which are invariably at least 6 db down from the peak if not much more (classic rock routinely has peaks 30 db over the 'body' of the sound as expressed in RMS loudness). This easily raises the grungy one-bit noise floor level of 16 bit CD audio up into the plainly audible, and as for any lossy-encoded format, forget it: an order of magnitude worse.

CD quality is the very least you could want for a serious big club or theater system (much less auditorium). Between peaks and the requirements for deep bass, the peak in a digital audio file is (a)much farther above the body of music than you'd think, and (b) should never be reached, because that's clipping.

People routinely behave as if the theoretical maximum dynamic range of a Red Book CD is relevant to anything. It's incredibly easy to play back music loud enough that the noise floor will get obnoxious and relevant to listening, it's only 96 db down. Any small system in an enclosed live space can push louder than that. Cranking music over headphones will blow way past that and you won't even hear the peaks, but you'll be able to hear how bad the subtleties (or lack of same) on 16 bit red book CD are.

Electronic music, especially live, is totally relevant to high resolution audio. I more than suspect some of the big name acts (for instance, Deadmau5) are using converters to the mains running at not less than 24/96. Certain synthesizer mixes are very revealing of faults in the playback. If the live performance over a modern PA sounds absolutely huge and awesome, but not strained or grainy, then they're not using CD quality. The SPLs are more than enough to make these distinctions obvious.

Anyone can get these SPLs over headphones, trivially, and headphones that'll handle it cleanly are only a few hundred dollars.

Doesn't this mean going past the 85 dBA limit in headphones? That's instant hearing loss.

I had satellite radio for a while but had to cancel it because I couldn't stand the sound of cymbals at whatever bitrate they encode at. Weirdly, when the phone operator asked why I was cancelling and I told him "audio quality" he acted like he had never heard that before.

I'm not an audiophile at all but satellite radio sounds worse to me than anything other than AM/FM radio. Spotify over LTE and Bluetooth sounds night-and-day better.

When I was considering Tidal vs. Spotify in the past, I ran across the ABX tests here:

http://abx.digitalfeed.net/list.html

Pretty much found the same thing for myself at 256+.

I only ever recall finding one track where I could hear differences in bass soundstage / positioning on 256+ mp3, vs. uncompressed 44/16, back when I was using stereo subwoofers in a heavily acoustically damped room. Even then, I could only tell when abx'ing.

You can't tell the difference on the vast majority of music. But encoders occasionally mess up, producing an audible difference. In the case of MP3 this often takes the form of a pre-echo. For example encoding cymbals and hapsichords seems to be difficult, though much less of a problem than it used to be. My understanding is that such killer samples exist for good MP3 encoders, even at 320kbit/s, but it has been a couple of years since I last looked into the topic.

(Personally I downloaded some of those killer samples and couldn't tell the difference, but other people reliably tell them apart in an ABX test.)

> he says he could tell the difference between high bitrate mp3 and 16 bit 44k wav.

In the early days, when all mp3 encoders were pretty bad, I could tell which encoder produced a given file. mp3 encoders today are vastly better. I've not been able to do that party trick in a long time.

I remember the Xing encoder was blazing lay fast, but sounded worse than LAME. This was around 1999, IRRC.

Yep, especially bad on cymbals - always a dead giveaway.

Yes and hihats. But how am I to tell now that I have high frequency tinnitus?? Blasted drummers.

And whenever someone said an 's'. It sounded like 'sshssh'

thank you for clarifying! It seems I slightly misrepresented your views...for that I'm sorry...I was going on memory of what I remember you saying.

> The source of the recording is usually the best master known for that record (special remastered editions, etc.).

Usually what is happening here is you're getting a master that hasn't been compressed to death (see loudness wars). Vinyl is a shit source for 'quality', but most records aren't compressed to death so they can still sound better on a good set of speakers due to the dynamic range.

So, in other words: bite the bandwidth bullet and get the 192 material, but for Pete's sake, do yourself a favor and downsample to 44.1.

If you have the patience, sure. In my desktop I keep everything on the big file source and listen to it that way. I also have dedicated DACs/AMPs that can decode all the way up to DSD. I know it probably doesn't make a difference but whatever, storage is cheap. For my phone I listen to spotify.

Similarly, seeing 24/192 on a rip from an analogue source (e.g. vinyl) usually implies that a high-quality player+amp+ADC was used to digitize the album.

It seems to me what we really need then is some other kind of litmus test that indicates a quality recording, without having to waste a ton of bandwidth and storage to get it. Are there other things to look for that also tend to correlate with quality? Newer formats maybe?

so like, signaling?[1]

[1] https://en.wikipedia.org/wiki/Signalling_(economics)

The 192 is really for low latency not for anything you can actually hear. You can't hear the difference past 48khz. An audio interface that can do 192 gets you down to 5ms latency where you can finger drum without being thrown off. But it has nothing to do with audio quality. Like if you use guitar rig you will have the lowest possible latency for realtime computer generated guitar effects at 192.

Where do you get the 5ms from? Why is 192 more likely to have lower latency? I would think the higher bandwidth required by 192 has the possibility to slow your computer down and cause more latency.

I believe the parent might be referring to the myth about attack of signal falling "between the cracks" of digital sampling. This is debunked in the video linked elsewhere, in the chapter titled "bandlimitation and timing". I might be wrong about the parent poster's meaning, though.

See the excelent answer by zaxomi for another more favorable interpretation of the parent. https://news.ycombinator.com/item?id=15131280

I believe the parent is referring to the latency of real-time effects used during performance:

* With analogue equipment the latency between the input and output of the effect is often below 1 ms (unless the effect is supposed to have a delay).

* Standalone digital effect equipment that processes the samples one at a time can also have a latency below 1 ms. (48kHz ≈ 0.02 ms)

* If you use a computer for real-time effects, the samples are not transferred sample by sample from the audio interface, but in a block of many samples. The number of buffers and the size of them can usually be changed by the user. With a buffer of 1024 samples, the oldest sample is already about 21 ms old before the computer can process it. After the buffer is processed it has to be transferred to the audio interface, and that will add another 21 ms. So the minimum latency for any real-time effect in the computer is about 42 ms at 48 kHz if the size of the buffers is 1024 samples. Often it is much worse because the operating system adds more latency. If the equipment can handle a sample rate of 192 kHz, the same latency is about 10 ms. If the computer can handle smaller buffers, the latency can be lowered. With 256 samples per buffer the minimum latency will be about 11 ms at 48 kHz and 3 ms at 192 kHz.

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