24/192 Music Downloads and why they make no sense (2012)
people.xiph.org
people.xiph.org
[2012] https://news.ycombinator.com/item?id=3668310
[2014] https://news.ycombinator.com/item?id=8689231
>Think it's 2040, singularity reached. AI runs the world and on HN we have this article popping up very frequently like every hundred Planck time unit.
One argument I can see in principle for 24/192+ sound (not music) recordings would be if someone was a serious transhumanist and honestly did anticipate that some humans will move beyond baseline human sensory limitations in the foreseeable future (by 2040 would certainly count). Combine that with the sort of incredible environmental destruction we're seeing right now, with enormous numbers of species going extinct, forests being destroyed, insect/bird levels plummeting/moving even if they aren't going extinct entirely, etc. It doesn't seem entirely unreasonable to imagine that in 2040 somebody with genetically enhanced or bionic ears who really could hear ultrasonics (and had grown up with that, so their brain had developed from the start with that input) would find themselves not being able to ever hear "what it was really like" back in the 2010s even for a simple walk in the woods. If they had been here in person they'd be able to hear all sorts of things, but our standard recordings wouldn't have any of that, and in that time the whole character of forests may be different forever ala the silent spring. It's similar I think to one of the obvious guiding principles of modern archaeology, which is to try to disturb as little as possible precisely because we recognize there will be superior tools and sensors in the future which could pick up things we can't right now. Saving as much raw data as feasible in many experiments is also like that, even if we can't process it all now decades down the line new insights might be found.
None of that has anything to do with music which is a subjective human artistic creation. Even though instruments give off sounds beyond our perception, by definition we aren't taking those sounds into account in the creative process. Future transhumans would undoubtedly create transhumanist art taking full advantage of any enhanced senses, but that wouldn't apply retroactively.
True, except that few microphones provide a useful signal over 20kHz, and in the case of produced music, that segment of the signal was never heard or "signed off" by the original artists/engineers and therefore can't be considered part of the artist's intent.
[0] https://xiph.org/video/vid1.shtml [1] https://xiph.org/video/vid2.shtml
And twenty years from now it's going to be hard because you'll have to scrounge the gear from a museum instead of it being available for a reasonable price from eBay or borrowing it off somebody who kept it in the cupboard after upgrading to modern digital gear. So I'm glad Monty did it in that era where the gear was still available.
It puzzles me that many people don't yet know about Opus. Let me quote the FAQ [3]:
"Does Opus make all those other lossy codecs obsolete?
Yes.
From a technical point of view (loss, delay, bitrates, ...) Opus renders Speex obsolete and should also replace Vorbis and the common proprietary codecs too (e.g. AAC, MP3, ...)."
[2] http://www.opus-codec.org/comparison/
[3] https://wiki.xiph.org/OpusFAQ#Does_Opus_make_all_those_other...
I.e. in practice - in my main archive I use FLAC. On some portable players and etc. I use Opus encoded from that FLAC.
That's why I always try to buy music in FLAC when possible and stores like Bandcamp are great for it.
Just a guess but I bet it's because the cost would be higher than the extra revenue it would generate.
I'd imagine they consider what they have is good enough considering the backwards compatability issue it'd likely introduce.
https://community.spotify.com/t5/Live-Ideas/Music-Use-Opus-C...
https://www.tate.org.uk/art/artworks/klein-ikb-79-t01513
https://contemporaryartetc.wordpress.com/2007/09/13/fact-of-...
But within the visual spectrum but not showing on the screen is still within the visual spectrum. The article examples refer to infrared and UV+ for contrast, and that's entirely correct. Monitors displaying either of those would make no difference (well, beaming ionizing EM at your face raises significant concerns audio doesn't at any level) at any point. They're simply beyond human eyes period. It's an accurate analogy. Failing to reproduce something within human limits would be what you're talking about, but that's a solved problem and not something 24/192 offers you anything with.
When DJing, I often speed up or slow down a track I'm cueing in order to match the tempo of the playing song. So having 192 kHz tracks might be better (although usually you try not to change a song's tempo too far from the original anyway).
It's kinda like how there's advantages of recording at 8k, better cropping, supersampling, etc. But for the average consumer there's no perceivable difference between the pixel density of 8k footage and 1080p footage on their 7" screen anyway.
More important than sample rate is AD/DA quality. I'll trust a new high-end converter at 48 kHz than an old prosumer device at 192 kHz.
Plenty of the albums we love as listeners were recorded at 44.1 or 48. Plenty were recorded with absolutely horrendous equipment but played and mixed by professionals who created magic. MANY modern vinyl releases where people brag about superior sound quality are just the CD master in all its 16/44.1 glory remastered for vinyl. Little of it matters when the end result is special.
If the producer is planning to slow down the audio (and wants the ultrasonic components to become audible), then recording at higher sample rates makes sense, and the author doesn't address this; probably this is pretty rare in practice. You'd also need ultrasonic-capable microphones.
The much more common operation is to filter or amplify the signal, and for that, more bits per sample is better to avoid amplifying your quantization error. The author covers this in the "When does 24 bit matter?" section.
No, it's literally excluded from consideration in the article's title. This is about music downloads, not music production.
> while keeping pitches the same
All 192khz does is preserve higher frequencies. If you're keeping pitches the same, there's no advantage to using an extremely high sampling rate for your source material. The advantage comes if you're going to lower pitches.
(Note that some algorithms need higher sampling rates to avoid aliasing. That shouldn't be the case anymore, but if you're hearing a substantial increase in quality just going up to 192 khz, most likely one of your algorithms is faulty.)
(Note 2: I say "substantial increase" because some people can detect up to 27khz.)
I didn't really see a mention of this point in the article since there was no "So when do you need 192 kHz?" section, but in its defense, DACs, amplifiers, speakers, and room ambiance are all incredibly linear in 2019, so for music listening, most super-sonic frequency content doesn't turn into to lower frequencies. It does matter when you're using the very nonlinear Apple earbuds, but if you were doing that, you wouldn't care about audio quality in the first place.
In most sensible systems, super-sonic content should be filtered out before it has a chance of doing nothing other than risking the fidelity of the final output.
As for your quip about a 100 kHz sine wave sent through a guitar amp, what you'd be able to hear are the distortions and subharmonics which are below 20 kHz—and if they're desirable in the recording they would need to be captured as their sub-20 kHz components. Capturing the >20 kHz components will do nothing but make the sound wildly and randomly inconsistent depending on the consumer's system.
That's less than half an octave over the "traditional" 20khz limit. Even the 20khz limit is more of an average then a strict biological limit.
It also means that a sampling rate somewhere at 54khz is the "ideal" limit when trying to pick a sampling frequency that is completely transparent to everyone.
This is less than half an octave higher than the traditional 44.1khz rate, just 22% more data.
That's the thing that really drives me nuts about high sampling rates. The minute improvement really only needs a very slight boost in sampling rates, not 96khz or higher.
Link?
https://asa.scitation.org/doi/full/10.1121/1.2761883
A decade has passed and it would be interesting to know how many people have reproduced the results detailed in the abstract. I gave it a quick read and at first glance it looks like an impressive experiment:
Hearing thresholds for pure tones between 16 and 30kHz were measured by an adaptive method. The maximum presentation level at the entrance of the outer ear was about 110dB SPL. To prevent the listeners from detecting subharmonic distortions in the lower frequencies, pink noise was presented as a masker. Even at 28kHz, threshold values were obtained from 3 out of 32 ears. No thresholds were obtained for 30kHz tone. Between 20 and 28kHz, the threshold tended to increase rather gradually, whereas it increased abruptly between 16 and 20kHz.
20, possibly 22 bit, and 60 to 80K.
Given that people screw that up by failing to dither to fixed point formats, you could push it to 24 bit, which is a generally supported word length. Since multipliers of common lower sample rates (44.1 and 48) give us 96K, that is also a good 'extra padding' to be certain of never encountering an issue.
I'm with Dan Lavry w.r.t 192K being unnecessary. Done properly, 96K gets everything, including extreme phenomena or artificial sound (for instance, I have a Farfisa organ that's capable of producing reedy thin sounds of extraordinary clarity, from simple electric tone generator circuits). I use 24/96 for my music stream recordings, while also streaming to YouTube at a much lower quality.
Though as others have noted CD quality won't improve a terribly mastered recording from the loudness wars.
I know very little about audio but my best guesses are:
1. The media cable was poor quality and/or playing music through the headphone jack is worse quality than radio station airwaves.
2. Spotify was sending back poor quality audio, possibly because I was not on wifi.
I'm sure the particulars matter but does anyone have a best guess as to why the quality would be so much worse? I don't really expect mainstream radio stations to serve up the highest quality audio, but maybe my assumptions are way off.
Amazon Music seems to be pretty good as far as quality is concerned. I think they download the MP3s onto the phone's local storage so they don't have bandwidth issues? Either way, I could hear the difference between Spotify and Amazon music. The difference between Amazon music and my own MP3s was not as apparent.
Pandora seems to sound "fine", although I seldom play it loud enough to notice. Spotify was the only one where I noticed the quality being notably bad. It's possible it's due to a low bandwidth fallback. And maybe they throttle their own servers at peak times, in addition to detecting the lack of local wi-fi.
When I last moved, I plugged my phone (running Spotify) into my receiver to check that I'd gotten my speakers set up right. It was so muffled-sounding that I was worried I'd somehow damaged my speakers!
4. You have a high definition radio and were listening to a high quality digital signal over FM as opposed to an FM analog signal.
It's probably a combination of all of these.
It destroys a lot of subtlety and sonic detail in the original, but in exchange you get an overall louder, more in-your-face sound, with highs that come through even on bad audio systems. On car stereos, where you have a lot of low-frequency rumbling sounds, this especially makes a difference. And if you ask a random person to give a subjective quality assessment of original vs that processed audio, they'll almost always feel as if the latter is of higher quality.
For more info see e.g. [1].
[1]: https://www.soundonsound.com/techniques/multi-band-compressi..., section "Broadcast Applications for Multi-Band Compression."
Why have an engine in my car that can exceed all speed limits?
Why have a heating and cooling system in my house that can exceed any comfortable level?
Why have lights that get brighter than I need?
Why have an internet connection that exceeds what I need now? ========
I keep all my music rips in uncompressed FLAC - 1) because i can 2) because I have the most flexibility (transcodes) 3) because it is capable of capturing _more_ signal that the original contains
No point in bottlenecking my audio just because _other_ people are unable to appreciate it.
The entire point of the post is that _nobody_ can appreciate it. It is entirely a waste of space at best, and a cynical marketing ploy at worst.
> Why have an engine in my car that can exceed all speed limits?
So I can drive faster than the speed limit if I want to. (And I do)
> Why have a heating and cooling system in my house that can exceed any comfortable level?
Well, you shouldn't oversize your HVAC system if you want to save money. But it's nice to be able to achieve your target temp in a reasonable time period. Any system that can heat your house by 10°F in 20 minutes will—as a side effect—also be able to heat it to 90°F if you were to set it there.
> Why have lights that get brighter than I need?
Other people may need that extra brightness. You can choose dimmer lights if you want. In any case, there's a clear difference between the two choices.
> Why have an internet connection that exceeds what I need now?
Again, other people may need that extra bandwidth. If you can choose a slower one, then do so.
The point of this article is that 24/192 downloads do not improve anything. It's like having a car engine with blue anodized cylinder heads. Nothing about the performance will benefit from the color change of the heads. Or using gold plated ducts for your heating system. The quality of the air is not affected by that.
Our ears are not capable of hearing the differences when they affect only frequencies above our range. Imagine if those lights boasted that they rendered 200nm light more faithfully. That improvement is wasted on your eyes.
It's like printing your brochures at 160,000 DPI instead of 2,400 DPI. The difference is entirely imperceptible by the human sensory system without artificial augmentation.
It's like capturing the invisible infrared light spectrum in a cinematic movie camera so it can be projected back to cinemagoers as infrared light in the theatre.
At 96KHz and higher with certain samples I can slow down by 80% and it will still sound good.
But if you mean just changing the speed of the sound, than you need to change the algorithm you're using. There should be no difference in quality due to sources having different sample rates.
This way, those who wish to hear how the music was intended to sound, will have a somewhat decent chance of coming near to what it sounds like, and people who want other flavours can still simply buy equipment which colors it in the direction they desire.
Perhaps we could give a bit of extra headroom for kicks, to widen the envelope at extremes however. A useful amount would look more like 20/48 rather than quadruple or sextuple the resolution. No one produces in this format though, the next one up is typically 24/96.
That aside - the bit depth part of this article is silly and wrong. With an unprocessed acoustic recording, the difference between 16-bit and 24-bit sources is fairly easy to hear on professional equipment.
By the time rock/pop/IDM/etc has been mixed and mastered, the dynamic range can be so limited you might as well distribute it at 8-bits. (Barely an exaggeration, BTW.)
This is not even close to being true of jazz, orchestral, and folk recordings. Typically recording engineers allow somewhere between 10dB and 20dB for peaks, which means the actual recorded resolution of sustained non-peaky instruments and quiet sections is somewhere around 12-bits - comfortably low enough to hear quantisation errors, even with dither.
So for some genres, 16-bits is plenty. For others it's nowhere near good enough.
In 2019, there's really no practical reason not to distribute music as 24-bit FLAC for high-end use. If you're listening on mobile you may as well use one of the better compressed formats. But for home playback, 24-bit is master-tape quality with no significant downside.
Sampling rate is a more complex issue. 48k is significantly better than 44.1k for the reasons mentioned.
Vinyl can go up to 100k or so, although not very accurately, and some people - including some very highly respected professional audio equipment designers, like Rupert Neve - believe that makes a difference.
But it's very hard to record ultrasonics "just in case" because the microphone->preamp->ADC chain has to handle them accurately, and that rarely happens. So there's very little of value up there in most recordings anyway - although maybe more on vintage tape masters than on modern digital recordings.
Personally I'm equally happy with 48k or 96k. The 192k recordings I've heard have been disappointing, possibly because of the intermodulation effects, but also because jitter becomes more of a problem at high rates.
It's neither silly nor wrong—the article's title literally excludes it from consideration. This is about music downloads, not music production.
e.g. https://pitchfork.com/reviews/albums/earl-sweatshirt-some-ra...
ffmpeg -i foo.flac -ar 96000 -acodec flac bar.flac
If you still think it's a problem, adding good dithering with ffmpeg's quantizer/resampler flags will make the noise floor 6-10dB smaller.
The article (and numerous other sources I've seen over the years) disagrees with you so I'm curious why you're so certain?
"It's true that 16 bit linear PCM audio does not quite cover the entire theoretical dynamic range of the human ear in ideal conditions."
Now, 24-bit may be overkill, but 24-bit is the next step up from 16-bit among standard encoding formats, and as the article notes, there are no drawbacks with 24-bit encoding except greater use of disk space.
"[...] does not quite cover the entire theoretical dynamic range of the human ear in ideal conditions."
Note the words "theoretical" and "ideal".
In your post it sounds like you're claiming that you can regularly hear a difference under normal listening conditions - which contradicts my reading of that sentence.
My gut feeling is that the difference you're hearing is placebo.
To put it another way - either the article is making an inaccurate statement, you're mistaken - or you've got golden ears and only ever listen to music in specially prepared environments.
Monty's gotta monty, and this argument has been going on from the very earliest days of digital: back when people behaved exactly the same way over digital recordings that are now commonly accepted to be excruciatingly bad for a variety of reasons (generally having to do with bad process and wrong technical choices).
You can get a HELL of a lot out of 16/44.1 these days if you really work at it. I do that for a living and continue to push the boundaries of what's common practice (most recently, me and Alexey Lukin of iZotope hammered out a method of dithering the mantissa of 32 bit floating point (which equates to around 24 bit fixed for only the outside 1/2 of the sample range, and gets progressively higher precision as loudness diminishes). Monty is not useful in these discussions, nor is anyone who just dismisses the whole concept of digital audio quality.
I believe it's a combination of imagined differences and barely perceptible differences elevated to implausible heights of significance.
Even if one can hear the difference between 16 and 24 bits it will be almost imperceptible in most listening conditions and when it is perceptible it will on the threshold - and certainly too subtle to affect the quality of the experience in any meaningful way.
96dB is a lot more than you probably think, it's like the difference between an anechoic chamber (nominally ~0dB) and someone jackhammering concrete right next to you (~90-100dB). Add to this that even a quiet room has a noise floor around 20-30dB, and to even hear the noise floor in CD quality audio, a full-scale peak would hit 130dB!
Try generating a sound at 0dBFS, the attenuate it in steps of 10dB and make note of when you can't really hear it anymore. At -50dB the sound is already extremely low and barely audible, and there would still be 46dB of attenuation available.
In addition to this, noise-shaped dither can push the noise floor towards frequencies where the human ear is less sensitive, giving a perceived noise floor of around -120dBFS. In other words, 24-bit audio for distribution and listening is absolutely pointless and has absolutely no audible difference when compared to 16-bit audio.
I did a lot of torrenting back in the 2000s, but thinking back on it I spent a ton of time finding things, organizing my file system, transcoding, editing metadata, etc. I do not miss that hassle at all now.
Personally, I buy music rather than torrent, but the pace at which I buy new music (either on bandcamp or physical CDs) costs me about the same as a Spotify premium subscription anyways, only I get to keep the music forever.
To provide you with another answer, most of the artists I listen to aren't on any of the music streaming services. Because local underground bands whom only have CD's handed out at their shows rarely exist outside of the pirating scene - which has a knack for distributing local underground bands with limited release/number of CDs. A small percentage of the bands/artists are on Spotify or Bandcamp but most aren't.
I buy what I can because I enjoy having the album arts but most of my music cannot be purchased or streamed.
There's also no guarantee that the streaming services will still exist in 10, 20, 30+ years - but there is an almost 100% chance that the hardware and software necessary to listen to or convert .flac will exist for me to continue to listen to my music.
The reason why a lot of recent digital music sounds bad is because of the intentionally terrible mastering. Since everyone is listening of crappy earbuds, they compress the hell out of it and destroy all dynamic range. This is why when downloading music you should avoid remasters (there are some exceptions, like the Beatles mono and stereo boxed sets that came out awhile ago) and go for the first edition presses.
This is also why modern vinyl releases sound a lot better than digital: they are mastered differently since its assumed everyone is going to be listening on good equipment.
That being said, I think flac is generally a good choice for a music collection. You can’t transcode mp3s without killing the quality so if you ever want to convert formats (like for a mp3 player), you should stick with flac (16-bit, 48hz).
The original idea of 24-bit 192hz flac was for vinyl rips, where hypothetically you might be getting more information.
I found myself to buy an iPod in... like... 2011 or so. Converted all the CDs I had to FLAC because losless was the way to go.
Two or three years (let it be 5, doesn't matter) pass by, I got a better Smartphone, Spotify Premium and don't touch my 1xx GB of FLAC music anymore, because I don't want to carry around another device etc.
I'm not sure but I think "owning" music like in "I got some files here on my drive" seems dead to me. That obviously has downsides but I feel lucky to use Spotify these days and being able to discover new music every day and listen to all of it on the go without buying something, converting it and more.
I still have Spotify for the times I want to listen to something I don't own or want to listen to one specific song without drilling down multiple menus to find it.
the loss of quality from transcoding lossy to lossy is usually a lot worse than the difference in quality between codecs and bitrates (within reason).
ffmpeg and a makefile with a pattern rule is pretty reasonable; (substutite any make-replacement if you prefer). If you are doing AAC, make sure you use the Fraunhofer FDK AAC not the builtin one (the builtin one used to be terrible, but is now somewhere between "okay" and "pretty good" but the FDK is still considered better last I checked, and your distro may not have an up-to-date ffmpeg).
ffmpeg is pretty good about preserving metadata.
If you want ID3v1 tags for MP3 (only needed for older players), then pass -write_id3v1; there's little downside to putting the id3v1 tag on there as it's quite small.
Links for basic ffmpeg encoding; it shows with .wav input but ffmpeg can read flac just fine and should preserve tags: 1,2
For Ogg output, oggenc can read flac directly and preserve tags, so I've never tried using ffmpeg.
I however, ripped my CD collection to a single flac per disc plus a TOC, and abcde[3] will automate that, including a musicbrainz or CDDB lookup for tagging.
1: https://trac.ffmpeg.org/wiki/Encode/MP3
On Mac, XLD is great for ripping and transcoding, but I'm not sure what's the hot favourite for playback these days.
Converting etc. I do exclusively on my Linux desktop, so can't help you there.
Your format of choice should be dictated by your mobile platform - if you use iOS device or simply like iTunes, go for ALAC. Any decent player will handle FLAC and ALAC, but Apple requires ALAC. If Apple isn‘t a concern for you, there‘s no reason to use anything but FLAC.
Personally, I use ALAC since I use iOS. So far there haven‘t been any downsides.
Absolutely, but it's an extra step that to me brings little practical benefit, since FLAC is already the source format & is more widely used practically everywhere outside Apple's ecosystem.
> Your format of choice should be dictated by your mobile platform - if you use iOS device or simply like iTunes, go for ALAC. Any decent player will handle FLAC and ALAC, but Apple requires ALAC. If Apple isn‘t a concern for you, there‘s no reason to use anything but FLAC.
I use iOS as my smartphone platform for now, (waiting for the Librem 5), but Linux on the desktop, so that's why I prefer FLAC. It's worth noting however that iOS itself does support FLACs perfectly well, just iTunes doesn't, (I prefer not to deal with iTunes at all, so not a concern for me), but if you use something like Airsonic, you're set.
I do have a set of AirPlay speakers however, since I wanted something wireless, but still lossless, which kind of means AirPlay is the only option & that does transcode my FLACs to ALAC on the fly, so there's definitely an area where I use ALAC, even if indirectly.
I mostly use 7digital & HDTracks to acquire FLACs these days, but when I rip from CDs, I use https://github.com/whipper-team/whipper to do the job.
FLACs from 7d/HDTracks are already named & tagged properly so I only deal with it occasionally and when I do, https://picard.musicbrainz.org works well for acquiring tags & artwork.
When I need to rename/tag manually, https://kid3.sourceforge.io has been working nicely.
Also I haven't used it myself, but there's a lot of positive chatter around https://github.com/beetbox/beets for tagging etc. I just prefer not to have my files touched in such an automated way :-)
I rarely actually convert from FLACs these days, since I have set up Airsonic, (https://github.com/airsonic/airsonic), on my home server. I now have access to the lossless files directly, from anywhere.
When I do convert, I usually just use https://github.com/kassoulet/soundconverter - nothing fancy, but does the job. I do not maintain my whole library in both, lossless & lossy formats since I have set up Airsonic, but when I do want to save data & do not have access to WiFi, I just let Airsonic use lame to transcode to MP3s on the fly, (rare). If you cannot do that, don't have regular access to data on the go etc. I'd honestly just use https://ecasound.seul.org/ecasound/Documentation/examples.ht... and put it in a script that checks if a .flac file in a folder or subfolder has a corresponding .mp3/.ogg file and convert if not, then just use find to filter out the format I don't want to copy over. :-)
Over the years I've ripped my CDs maybe 4 or 5 times. I used to have a PowerBook G4 and an early iPod, so I ripped to M4A/AAC. Nothing else played that, so then I went MP3 with storage limitations of the day dictating bitrate. Now, I just want to rip to FLAC and never deal with that again.
I keep the Flac around in case sometime in the future I want to change formats for whatever reason.
I use my iPod Shuffle exclusively for portable music listening. Cannot beat the form factor, only have to charge it once a week or two (and sometime far longer between charges), and helps me relegate my mobile surveillance/communications device to phone-duties-only as much as possible.
Nowadays, I also just use spotify since I don’t have a quality source for music. But if what.cd was still around, I would dump spotify in a second.
Spotify is fine for casual listening, but if you're picky about quality, you're going to diy it, and if you're diying, 24/192 is pointless.
CDs? People with a room full of 8 tracks or cassettes would like to have a word.
HDDs? Those fail all the time, plus any sort of natural disaster could wipe out your collection.
Online backup? This seems like the only real option, but for me the risk/reward just doesn't fit.
At least for now, the record companies and the service providers are both incentivized to have as much of their catalogs as possible on streaming services. Until that changes, streaming works for many.
What risk? You can privately store your music anywhere, it's completely legal to do so.
Preferably not hosted in the US, for privacy/bandwidth reasons.
For now, I've decided on pCloud, in addition to an on-site copy on my NAS and a copy on a portable drive that I store at work and update semi-regularly. A couple of rsync scripts take care of everything.
I know it's a cloud storage service and not a an actual proper backup service, but they offer 15 days of rewind as standard, and you can get a full year of rewind as an add-on, which I am considering. That should hopefully protect me from accidental deletes, and give me enough time to restore if my house burns down.
The thing that has really sold me on pCloud is that their Linux client is absolutely amazing. Compared to the barely functional Dropbox client and the non-existent Google Drive client[¤], it is an absolute joy to use. At the moment it's an Ubuntu-only AppImage, but they're working on an improved Electron version.
One additional nice thing is that pCloud is a Swiss company, so their privacy laws (and the GDPR) apply. They do host their servers in the US, so you're not completely free from theoretical NSA/PRISM snooping, but in my case I'm primarily storing my music library. They can go ahead and snoop through the tags of 300+GB of music for The Anarchist's Cookbook or whatever.
[¤]InSync is pretty nice, and I did buy a license for it a while ago, but it's still not as good as pCloud's client.
I can access this from every device in my house, and from outside my network.
I can put anything I want onto my phone, USB stick or iPod and play in most any modern car.
-My desktop at home
-My server in the basement
-My work laptop's external hard drive
-An external hard drive in a fireproof lockbox (server backup)
-An external hard drive on a shelf at work (server backup)
-An external hard drive in my parent's house 150 miles away (server backup)
Try prying my files from my cold dead hands.
Luckily it's on a backed up RAID6 array, in private server, streamable whenever I want.
I mirror all my purchases onto equipment I own, and so I guess I get the benefits of both.
There is a solution for the rest: let me mix songs from Spotify, my own library, and any other services I pay for in a single playlist.
- stored on the desktop for fast and performant access
- synced to an NAS daily for central access around the house/network
- uploaded offsite to cloud storage daily as backup
I really don't think that's true. I think the "listening market" looks a lot like it did before; a large number of casual listeners and a smaller number of people who are in to their music enough to care about details. The second category does things like talk about differences in mastering between different releases, for instance, and Spotify or Apple are not going to offer you that 1973 Berlin recording or whatever. Tidal tries to cater to this market, but they don't have a massive amount of stuff. And then you get to bootleg collecting and people who record performances, old music that didn't make the digital jump and all sorts other recordings that will never make it commercial services.
I'm not a "real audiophile" or obsessive about collecting things, but I do have a lot of music (last I looked, about 60k distinct artifacts - mostly individual songs, but some of those are albums or nonmusical, also some dupes and garbage). And a lot of that is not on commercial services.
In addition, I find that I use the MP3 player when I'm out running normal errands precisely because I've organized my music by hand and even edited tracks by hand in some cases. Examples would be things like rare covers that can only be found on YouTube, or favorite songs from niche internet music communities which were poorly mastered.
It's also a bit of a gear hobby now since there are so many MP3 players on the market. Prices are low and performance is great.
I have to agree about the iPod though, as I found the need for proprietary software, and really annoying software at that, made me use it less and less until my 32GB iTouch was mostly used as an ebook reader. I also prefer physical buttons for my mp3-listening while on the go.
While I do also have a Spotify Premium subscription, I am using it a lot less now than I used to. At least 10% of the album's I have simply aren't available on Spotify, and possibly never will be. Underground self-released artists very often don't bother with streaming services, or are outright against the entire concept in the first place, claiming that it devalues the music. It certainly doesn't pay very well. There's also the issue of music disappearing because of rightsholder disputes, such as most of the Motörhead discography being unavailable for an extended period of time. That sort of thing just isn't acceptable.
Honestly I've come to realize that I prefer a smaller nicely curated collection over a massive unwieldy semi-unlimited library, with questionable curation. I have reported hundreds of curation errors to Spotify, but they keep popping up, especially errors involving two identically-named artists being mixed together.
I will admit that I am very particular about tagging, labeling and sorting by genre. Spotify is woefully inadequate in this regard. For my own collection, I am in full control, which makes it much easier to sort and handle.
But this is kind of a pedagogical example. Not the point of who you're responding to.
[0] https://en.wikipedia.org/wiki/Dual-tone_multi-frequency_sign...
With older encoders, cymbals were terrible, but lame's psychoacoustic model is pretty good at masking those artifacts these days (at least at high bitrates).
For examples, some quick searching on hydrogen audio found a couple songs reported to be ABX distinguishable with lame and after previewing the songs, they do in fact have a lot of quick attacks:
Human Disease by Slayer (Some very fast and cleanly played drum parts; also it's mostly not snares; snares sound terrible at low bitrates, but I personally can't distinguish high-bitrate snares from uncompressed snares)
Show Me your Spine by PTP (The "instrument" used for the base rhythm has an unnaturally short attack).
I don't really hear the pre-echo.
Actually I love that I don't hear compression artifacts in music or see them in JPEGs. Makes my gear so much cheaper. :-)
More compression and less dynamic range is beneficial for certain environments. Noisy subways. Watching TV in a noisy downtown apartment. Basically, crappy, noisy environments. In those, compression will help you actually hear the music and speech. However, the fact that this should be done in the master is an artifact of an earlier time. Now that signal processing is small and cheap enough to be ubiquitous, music should be mastered for the best equipment, then appropriate signal processing should be done by playback.
The problem is, that there is a lot of older equipment out there that wouldn't be able to do this. So the signal gets compressed before distribution, as a compromise for the least common denominator of equipment out there. Otherwise, a big chunk of the population would think the master sounds like crap. To them, in their particular situation, it would.
EDIT: Come to think of it, the current system, where most music is more compressed, but where the people who care can still get a high dynamic range version, is a very good compromise. The problem is that the latter group's selection isn't quite filled out by the market.
So I think it makes sense that records are mastered with a lot of dynamic range, so the people who actually enjoy music can enjoy it, and the people who don't can just equalize it themselves.
So I think it makes sense that records are mastered with a lot of dynamic range, so the people who actually enjoy music can enjoy it, and the people who don't can just equalize it themselves.
You do realize that you just restated my comment, but left out the analysis of the current day situation? BTW, equalization doesn't directly change dynamic range. Equalization is meant to change frequency response. It can change dynamic range by causing clipping.
If you're generally talking about "Loudness Equalization" then in many cases, it really is equalization. I don't know about anyone else, bit I've been talking generally about loudness equalization the whole time.
Windows loudness equalization is not equalization, despite the silly name. Ironically, I imagine Microsoft specifically didn't call it compression because most consumers only think of the other compression. Good grief.
Well, you learn something new every day. In this case, it's yet another time marketers have completely diluted the technical meaning of terminology.
You attach some metadata to the audio file that says certain parts should be level boosted in a noisy environment and there you go.
Similar to the thing we did witb vinyl back in the day where we wanted to fit more music onto the thing and applyed the standardized RIA filter when cutting the template — every phono preamp reverses this effect.
The tbing is people need to have specs they can mix and master for. Making something up makes mixinf unpredictable and that is bad.
At lower volumes, we perceive mid-range frequencies to be more prominent than at higher volumes. The loudness buttons would add lows and highs and/or lower mids so that the music would "sound better" at lower volumes.
Good point, I think particularly for movies or such this makes sense. I want to be able to watch a movie such that I hear what the characters are speaking, without blowing my windows out of their frames during some action scene. Yes, I realize in real life explosions, guns etc. are really loud, and this makes the movie less realistic.
Really loud? Ear-damaging loud! When the realism becomes actually endangering to your health, your escapist media has gone a bit too far.
This means usually the digital -6db is the maximum loudness with a short term maximum of -1db and dialogue at -9db.
Music today should be mixed according to the EBU R128 today (at least in Europe and for radio) which is a serious win against loundess maximizers and limiters.
I’d argue for in decice DSP compression for small earbud people, and give the whole dynamic range to the rest of us : )
Microchips for leveling audio gain existed in the 1980's and were found in consumer equpiment like TV's.
For readers of your comment and your child comments, it is important to note that the compression you are talking about in that sentence is not the same as the compression that most people are thinking of when discussing digital file formats (mp3, etc.).
This might be helpful:
https://forums.stevehoffman.tv/threads/the-common-audiophile...
High resolution is absolutely important in some mixing scenarios to prevent pre-ringing and aliasing in the effects chain (distortion effects or otherwise). But once you have your hi-res master, there is zero advantage to distribute it that way. At that point, a 48Khz/16-bit FLAC is as good as it gets.
To say nothing of how generally available vinyl records (especially old ones) have wildly different rms/peak measurements than generally available CDs and digital recordings have. This is partly 'Loudness War' and partly vinyl's inability to even do the loudness war thing and cope with blocks of heavily limited audio in the first place.
So you'll end up with a record where you can play it, and the peaks are 30 freaking dB over the RMS and it sounds amazingly open and uncompressed… while there's also groove noise that is every bit as loud as the music is (admittedly annoying).
A person arguing the vinyl/CD dynamic range thing would make the claim that the record was equivalent to maybe TWO bit digital audio, or four bit. The most cursory listen to such a comparison will show how inadequate it is.
I agree that the quality of the record -- AND its playback equipment -- among other physical factors will dramatically effect the numbers. My "10-14" quote only applies for ideal conditions: a newly-minted, unplayed disc on a high-quality preamp which together with the turntable and clean needles can produce a very low noise floor. Obviously I'm never going to get this with my dad's old Dead vinyl that he played to death, or with cheap needles, or with those crappy Crowley turntables at target....
Anecdotally, on my home system with clean records, I can make nearly-CD-quality recordings, with the differences only really apparent on flat studio monitors or a good Hi-Fi.
Surprisingly, yes. With noise shaping (https://en.wikipedia.org/wiki/Noise_shaping), very coarsely-quantized digital audio can produce high signal to noise ratios in the audible frequencies, via quantization techniques that push the error towards ultrasonic frequencies.
This doesn't violate information-theoretic limits because noise shaping requires very high sampling rates. The 1-bit Sony DSD format (https://en.wikipedia.org/wiki/Direct_Stream_Digital) used a 2.8MHz sample rate.
In the case of vinyl, the effective sample rate is physically limited by the (linear) record speed divided by the vinyl grain size, and to a rough approximation the bit depth would be log of the maximum groove amplitude divided by the grain size. However, the analog cutting mechanism would greatly limit the opportunity for dithering and noise shaping -- for example a needle cannot cut a wave shorter than the tip size.
I did a blind test between 128-mp3, 320-mp3 and flac hearing classical music. While it's true that the 128-mp3 is obvious to find, it also isn't too difficult to find the 320-mp3. Flac just sounds better. Described as a feeling, flac is more voluminous and doesn't feel cut short. For fun, I also let my parents take this test and they could tell, too.
That's why I converted all our CDs to FLAC. Storage is cheap anyway.
The only artifact I can reliably hear in 320 kbps MP3s is pre-echo, for instance with castanets, and only in a few very specific situations. Apart from this, V2 and above sounds completely indistinguishable from the original to me.
but what if in the future you could hear the differences? And now your entire collection is in MP3 V2 - now what?
No reason to not rip everything in uncompressed FLAC these days.
But if your happy with your audio now, great!
And even though you may have diminished hearing you still have awareness of frequencies above 8Khz.
Imagined artifacts will probably remain or might even increase though, since people typically have much higher spending power at 40. :)
I worked in audiology for my entire 30's and tested my own hearing (biocalibration) once a week.
Never once saw a reduction in my hearing thresholds (250Hz - 8kHz)
Nor have I read any supporting documentation that agrees with a reduction of 8kHz thresholds before the age of 55.
https://www.sciencedirect.com/science/article/pii/S089662731...
Stick to FLAC and keep your fingers crossed for cochlear regeneration tech ;)
Other than that MP3 (or Vorbis or Opus, which would probably do better on that song) is great for portability, but I'd still use FLAC for storage.
Something also ends up missing in the midranges. I was working on a track once where all I had was 320 mp3 version of the vocals. At some point I replaced it with a flac copy of the same vocal recording, from the same original wav source and the difference was noticeable right away without changing any of my equalizer settings or anything. It just punched through more and the clarity improved.
I have a room-corrected setup with two properly adjusted subs, and MP3 does just fine on deep bass content.
Regarding solo vocals, the history of the MP3 format says: "The song "Tom's Diner" by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the scheme, making sure it did not adversely affect the subtlety of Vega's voice".
That's not to say that they did a perfect job, but human voice was a very high priority.
And the encoders have continued to improve. So an earlier encoders may have messed with the voices, but a reasonably recent version of LAME would do so much better.
MP3's real weakness is fast sharp transients, such as castanets and harpsichord in sparse recordings, where no other sounds can mask them. It's a fundamental weakness in the format, and cannot be completely solved.
Newer formats such as Ogg Vorbis, Opus and AAC do not suffer from this weakness.
Well for me it came from running multiple copies of the same bass heavy tracks encoded in different formats through spectrum analysers. But I guess those lie?
>Regarding solo vocals, the history of the MP3 format says: "The song "Tom's Diner" by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the scheme, making sure it did not adversely affect the subtlety of Vega's voice".
Human voices come in a wide range of tones and frequencies. Optimizing something for one voice doesn't mean all voices will benefit from the same optimizations. The specific track I was referring to had a lot of variation in high and low notes. You can tell me all you want what I did and didn't hear.
Of course it's going to look different in a spectrum analyzer, the whole point of lossy compression is to discard parts of the audio to save space.
You can't evaluate the quality of a lossy codec by looking at spectrograms. They're designed to fool human ears, not measurement software.
I had always assumed they were taking the same master and just carving it into vinyl. I wonder what percentage of "modern vinyl releases" are actually remastering before pressing...
Phone = LG V20 which has an [ES9218](https://www.androidauthority.com/lg-v20-quad-dac-explained-7...) chip for its DAC.
Headphones = Sennheiser HD380 pro, pretty good for under $200.
Soundcard = "ASUS Xonar DGX PCI-E GX2.5".
Sound source = FLAC, Google Play Music subscription)
I'd like to upgrade to a really nice DAC and headphone amp to connect to the PC via USB, but that's way down the list of spending priorities.
I know that I'd probably have trouble distinguishing between audio components and sources in a blind listening test, and of course I have tinnitus, but I think my current "setup" if you can call it that is good enough for most stuff.
I am absolutely with you on the loudness wars though. It's a joy to listen to stuff that has real dynamic range, but it's not something I obsess over when I'm listening to music in the car for instance.
There's nothing extra in a 192kHz signal that would help with the vinyl mastering process. You could make a technical argument for the benefits of a 24-bit source, but in practice even those benefits would be utterly swamped by the SNR of vinyl.
And you might even have a point, as long as you acknowledge that “remastering for vinyl” doesn’t actually necessitate distributing on vinyl, and what you describe as “the sound of modern CD audio” is entirely the fault of human decisions and not the CD format itself.
Also, you’d need to acknowledge that your description of what “sounds better” is a subjective assessment. It is fair to say that vinyl sounds better to you if what you like is that RIAA processed, variable noise floor sound.
Of course in practice I do still keep flac rips around because I'm a data hoarder and what if I decide I want to reencode all my music to opus or something? But at least I have the option to stop caring.
Releated: https://thevinylfactory.com/features/analogue-digital-vinyl-...
There are, of course, those brands that care about remasters, but I don't think they're a majority of the market unless you're looking at classical and older jazz.
[0] http://dr.loudness-war.info
There are of course multitude of other factors impacting mastering quality, but as far as DR goes, this DB is a pretty good source.
https://wiki.hydrogenaud.io/index.php?title=Myths_(Vinyl)#Ef...
I'm going to disagree here. They are mastered differently because the physical limitations of the media require them to be mastered differently _and_ it just so happens that the physical limitations help limit mastering tricks in a way that produces less fatigue-inducing, brick-wall-limited mastering output.
A heavily compressed master creates huge peak-to-trough cuts in the vinyl which can cause the needle to literally jump out of the groove, even with RIAA limiting applied.
The assumption of the gear is definitely not true in any mixing or mastering experience I've had. Mastering tries to balance the final product across a range of listening devices, not some unobtainable ideal system. NS10s are kicking around because they sound like arse and make for mastering results that work well on car stereos and other "inferior" systems.
https://wiki.hydrogenaud.io/index.php?title=Myths_(Vinyl)#My...
They spotlight midrange with a presence peak right where the ear's most sensitive, and this is in part because the woofer is actually designed more like a midrange: thin paper, conical rather than curved cross-section, both of which also contribute to 'sounding bad' tonally while delivering energy more unforgivingly.
They're not really about mastering, though, they're about mixing because if you have elements out of balance it will be screamingly, annoyingly obvious on NS10s. That's not down to their bad-soundingness, it's down to their ability to be incredibly unforgiving.
One other consideration for a music collection from CDs is getting a good rip in the first place. I've had some horrible rips in iTunes, even with error correction enabled. I have much more confidence using a tool like XLD that supports AccurateRip, which probably doesn't work with a lossy format.
If you want to transcode after the rip, fine, but you may as well hang on to the FLAC.
Sorry, I don't know much about sound so here comes probably the most stupid question of the day (but hope never dies):
does this mean that I might get better sound if I would buy a vinyl & one of those turntables which can directly digitize to USB, then if I would buy & download the digital song directly (or maybe even the CD)? Thx
Didn't the beatles famously create their music to be listenable on the terrible radios of the time?
Look, I actually grew up with vinyl and 4-track tape, and audio cassettes. Unlike most folks being all trendy and hip nowadays, I've years of using that stuff.
Analog is shit. It's noisy, has a ton of distortion, and it gets shittier every time you copy it. Oh, and if you just keep it in storage, guess what, it decays just by sitting there (vinyl collects dust and scratches when used, slightly different).
In 2002 I built my DAW (digital audio workstation) and recorded my first tracks in 24 bit digital. Zero noise, zero distortion, no generation loss. It was like alien technology.
Digital is better in every way, by a wide margin. Period.
https://www.reddit.com/r/headphones/comments/awyy1q/this_pre...
Current mastering practices prevailing in the industry make no difference on this matter. Analog is still garbage. Find digital copies that are mastered properly and you'll be fine.
That being said, I still only buy music in CD, due to all the hassle of DRM and playback. I just want to drop in a CD and listen to the entire album, not futz with computers, encoders, and software.
I have a simple CD player, kit built tube amp, and homemade single driver speakers.
That's what I do. I tend to favor old master copies.
I rip them to both FLAC and MP3. The former is for listening at home, the latter for mobile scenarios. I store everything on the Linux server at home, and share via UPnP. VPN into the home network gives me access from anywhere.
Foobar2000 is my preferred player on Windows, BubbleUPNP on Android.
> kit built tube amp
What's the distortion on that thing?
The vast majority of tube kit schematics are very old tech, stuff that engineers from the 1930s would recognize. Their THD (coefficient of total harmonic distortion) is very high. What is known as "tube sound" is basically just huge THD, along with a specific distribution of energy across the harmonic orders.
It's fun as a hobby, and for the satisfaction of building stuff on your own, but even the simplest schematics built on modern principles vastly outperform these things by essentially all metrics.
Some tube amps are built specifically for low THD, but unfortunately they are rare. When in doubt, use solid state.
> homemade single driver speakers
I used to build everything myself back in the day. Speakers and amps was just part of it. Also did automation, radio frequency (I'm a licensed HAM radio operator), digital circuits from logic gates to DAC/ADC to systems with microprocessors to small computers. It's a small miracle I didn't actually go into electrical engineering.
You really need multiple drivers and likely a subwoofer also, to cover the whole audible spectrum.
https://wallofsound.ca/audioreviews/amplification/review-ele...
I’m not worried about the whole spectrum as I only listen to classical on it, primarily string quartets. The speakers are folded horns, so bass response is reasonable.
Anything like Dream Theater or Iron Maiden is in the car. Would definitely be solid state with a subwoofer for that.
I guess that's why the vinyl versions of my wife's albums always sound better than the downloaded versions. Even to my really quite bad ears.
Basically, to my ears, it just sounds like a bunch of early reflection reverbs were added (an effect that was mature in the 1980's in its high-end implementations and used in studios to get "bigger" guitar sounds and whatnot.)
Of course, it sounds great for all the viewers who are using cheap (or even not-so-cheap) earbuds, or computer speakers.
What these nincompoops don't get is that these albums were made to be cranked up on a powerful stereo, with full sized speakers, in some kind of room. That guy is basically just ruining great albums who were actually recorded and mastered by people who did know what they were doing. Like, oh, Detonator by RATT and whatnot.
So, there is a case to be made for this kind of processing. But I won't trust a random mastering "guru" with unknown credentials to get that right.
Also, I'm a little bit surprised that nobody focuses on more "out of the box" perception of sound. One can absolutely sense hgh frequencies, personally feel kind of like pressure where you can't pop your ears to equalize. Playing around with this feeling adds emotional tension and color to tracks.
Also, interference patterns are perceptible, and they sound kind of... Different from pure tones, idk.
The loudness war isn't happening because of "crappy earbuds", the earbuds included with smartphones have been rather good for a long time now. The ones that came with my Samsung S8 were designed partially by AKG (Samsung owns the Harman Group, including AKG) and are really damn good. Apple's included earbuds are also very good now, a far cry from the original iPod earbuds, which were decidedly mediocre.
The real issue is radio and Youtube/streaming services from before they implemented loudness targets, and it's been going on since the 50s at least, just listen to some old singles from back then, they're mastered as loud as they possibly could, with the technology of the day. The objective has always been to make your song sound louder than the next song, because louder music sounds more impressive to a casual listener, it's simply more attention-grabbing.
In the beginning of the digital era, there was actually some hope that better dynamics would happen. In the guidelines for Sony's earliest digital recording equipment, the recommendation was to target an average level of -20dBFS, to use very little or no compression, and "let peaks fall where they may". Just imagine that, 20dB headroom!
In the worst days of the loudness war (~early 2000s) lot of music was mastered with barely 3-4dB of dynamic range, with peaks banging hard against 0dBFS. I have some CDs from that era, and they clip and distort like crazy, because everything was just pushed to 11, to be as loud as possible. "Californication" by Red Hot Chili Peppers is an excellent example, it's absolutely horrid.
Since then, two major things have happened to improve sound quality somewhat. Firstly the compression devices and plugins have improved massively, modern sidechain compression is really impressive, entire genres like EDM/dubstep simply wouldn't exist if not for the improvements in compression tech. Secondly, all of the streaming services use volume normalization now, with a set average sound level. Songs can peak over this average value, but the average must be in line with the target. This also results in brickwalled "turn everything to 11" tracks sound a lot quieter, because they have no peaks to use the additional dynamic range available.
In my opinion that‘s a myth and certainly not a given. There are plenty of subpar vinyl masters and terrible pressings out there. And it‘s not that difficult to find good digital masters these days. More important than the medium is the genre, label and target audience - I have a pretty obscure and diverse taste, including rarities from past decades which are finally being re-issued for the first time and while mixdowns certainly vary in quality it‘s mostly fine and the result of a careful process these days.
However things might be worse when it comes to mainstream music.
I have just downloaded "Radiohead - The bends" and "Smashing Pumpkins - Mellon_collie_and_the_infinite_sadness", both apparently from vinyl and in highest quality but I don't hear any difference from the CDs I bought and ripped years ago (using headphones "Beyerdynamic DT 770 pro" directly connected to a Lenovo P71 notebook).
Maybe you meant some more modern music or something else...? Thx
[0]: http://www.noise11.com/news/r-i-p-pono-neil-young-kills-off-...
The Pono is kind of 'polite' sounding, and possibly not the most mastering-grade accurate playback compared to far more expensive DACs, but by God it sounds good. If you come across any at pennies on the dollar you'd be a fool to not snap them up, it's an extraordinary little piece of tech. (if you do and you don't want them, give 'em to me!)
Sure, 24/192 doesn't physically improve the sound you are able perceive when listening to it.
But listening to music is a highly subjective emotional experience.
If a listener cares about getting the best possible quality listening experience and feels downloading 24/192 music will achieve that, then the listener will actually enjoy music more knowing it is playing from a 24/192 source.
Listening to music is all about the feels.
Of course, I get how this can be abused. Next thing you know someone will be selling 32/320 for twice as much, then 64/480 for three times as much, etc.
Not that this kind of article isn't still really important. It is. It provides a lot of reassurance to audiophiles that they can enjoy their music to the maximum without buying into the 24/192 hype.
And that's what it's really all about: the best enjoyment of the music.
I can't stand seeing frequency response charts and scientific measurements in articles about audio. Like my favorite audio reviewer says [1], I listen to music for enjoyment and I talk about audio in subjective terms like "warm", "lush", "wide soundstage" - not "unexpected 14.5kHz falloff". I don't go to a restaurant and demand to see pH tests or measure the temperature of my steak myself. I'm not going to do blind A/B listening tests because I don't care about that. If you told me you liked one wine, would it be appropriate for me to say "No you didn't. You don't have taste buds that can tell the difference between that and any other wine."? Of course not.
Music is an entirely subjective experience and trying to distill it down to data is both condescending and telling of how little an author cares about music. Even if you don't care about subjective experiences of audio, why are you so bothered by letting people like what they like? How does it affect your life that I listen to music encoded at 24/196?
I've never heard of what newaccoutnas is talking about, but I guess if a high-frequency sound is ridiculously loud, the energy could be absorbed somewhere in the cochlea. I've only heard of experiments of subsonic hearing damage, not supersonic.
Maybe not relevant in the context of GP's post though. In the context of digital tools telling you two signals are identical, then I suspect they are, and if you want to prove to me your ears hear better then you're going to need a blind A/B test.
>How does it affect your life that I listen to music encoded at 24/196?
It doesn't, but Monty contributes to the Xiph project, which constantly has to deal with thousands of people saying "why doesn't Opus/Vorbis support 192 kHz??" This is his answer to that general question. There is actual money on the line for companies that use audio codec technology, and supporting useless formats that have no psychoacoustic effect is a waste of time, money, and effort.
THD, SNR, frequency response and many other metrics are easy: they define either accuracy or precision. If you want more than that, add external affects hardware or DSP. The purpose of the amplifier is to amplify.
You seem to be highly inexperienced in the subject, THD, IMD etc are parameters that are measured on highly artificial signals - singke sine waves or mix of small numbers of sine waves. Real music signals more resemble white noise than clean sine waves, and there might be hundreds of different very subtle modes of distortion, that are very difficult to measure but which have very significant influence on the precepted quality of the sound.
An industry which is known to sell an enormous amount of snake oil...
Yes, we have instrumentation better than the meat-based transducers in the human body. That doesn't mean we completely understand said meat-based transducers or how that meat-computer in our skull interprets signals from those transducers. The auditory and visual systems are still subjects of active research with many outstanding problems.
I mean, I'll be among the first to call bullshit on audiophile snake-oil like people pushing 192 kHz/24 bit music, but in your rush to discredit such things, you've gone too far in the other direction.
Even if they’re ignorant, I don’t care how they enjoy themselves. The problem is when they start writing guides and reviews encouraging others to follow their ignorance.
Is that the case here?
[1] https://news.ycombinator.com/item?id=19320093 [2] https://www.youtube.com/watch?v=RlCG2fK-abo
I’m not really inclined to watch a 25 minute video on this. What’s the timestamp of the blind test?
Like music, coffee is an entirely subjective experience. Like music, distilling it down to data does _not_ implicitly destroy the experience. Judging a cup of coffee good or bad solely based on data is impossible. Maybe this cup is intentionally overextracted and that song is intentionally overmastered. That doesn’t guarantee that the coffee or song are good or bad, it simply helps you understand why you do or don’t like it.
After making a thousand cups of coffee with a TDS meter, I can predict what would make a cup of coffee better without needing the meter anymore, and I’ve learned that I care more about enjoying coffee than I do about perfecting it.
If you sat through a blind test of a thousand songs, and at the end discovered that you _can_ distinguish 16/44 from 32/384, you might still _choose_ not to care. Most people don’t want every cup of coffee to be competition grade because it’s really expensive (density sorting), really difficult (dual-wielding flow-restricted kettles), and the payoff isn’t worth it every day to them. Maybe that’s how you’d feel after A/B testing 1,000 songs, as I did after pouring 1,000 cups. Maybe not.
For most people, knowing that 16/48 is indistinguishable or better than 32/384 will save them thousands of dollars and hundreds of hours of audio setup, tuning, design, repurchasing, etc.
For a few people, it’s worth it to them to go competition, either in coffee or in music. That’s certainly their right, but it’s not at all guaranteed to make them any happier than they would be with 16/48.
Would you accept that argument from someone purporting to provide medical advice?
Given that, your rant seems misplaced, and I can't help but think engineering is not the field for you.
Why push back against 24/192? Because it's a solution to a problem that doesn't exist, a business model based on willful ignorance and scamming people. The more that pseudoscience goes unchecked in the world at large, the harder it is for truth to overcome truthiness... even if this is a small and relatively insignificant example.
And yet just yesterday I was reading an audiophile magazine review that discussed a pair of $4,000 headphones and how the "transition from midrange into treble had something of a falloff that was confirmed by their equipment".
Music lovers listen to stereo equipment to hear their music. Audiophiles listen to music to hear their stereo equipment.