It's a very good article which shows up again and again. 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 .
It is a good article, and since the misunderstandings are persistent in the same way a lot of other commercially exploited mysticism is it remains a relevant one as well. Having said that and to try to add on something new to these discussions, since you brought up this:
>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.
This reminds me of Monty's A Digital Media Primer for Geeks[0] and Digital Show & Tell[1] - the delivery, the explanations and the way the experiments are set up is superb.
The article's author, Chris "Monty" Montgomery, is one of the authors of Ogg Vorbis [1] and Opus [2].
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, ...)."
I use Opus for music playback for all my archived music. The reason it's not more widespread was opposition of the likes of Apple to free codecs. Today they are losing this, and Opus is making its way even to Apple's systems.
I love opus just as much as I loved musepack and vorbis, the one thing all of them lack to one degree or another is support and hardware acceleration. If I throw an opus file on my Android 8.1 phone, it has no idea what to do with it unless I manually open it with vlc or foobar. For the regular user the support needs to be seamless, otherwise they are not going to bother.
The use of analogue gear in #2 is one of those things that as someone who _already believed what Monty is showing here_ I wouldn't have thought to do. But it really heads off a bunch of arguments.
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.
Honestly it is remarkable how many engineers (self-proclaimed or otherwise) in audio don't understand the basics of sampled systems and quantization. You'd think that anyone making broad claims about these kinds of systems would have at least a rough understanding of the foundational principles, but no.
The choice of colour analogy is unfortunate, because there really are colours that are "out of gamut" and cannot be accurately reproduced on normal monitors. If you have the opportunity to and look at one of the IKB works in person you'll see what I mean.
I don't quite agree with you, taking into account it's an analogy designed to help illustrate the issue for general audiences. It's not as if we don't have ProPhoto RGB or other wide gamuts or don't understand the issues of rendition accuracy and resolution within the visual spectrum. There was never any debate that sRGB alone in particular was quite limited, or that dynamic range was an issue. It's just that it represents a ton more data and is technologically and commercially much, much harder. As tech has caught up displays have continued to chase human visual limits, starting with resolution, then frame rate, and finally major industry wide improvements to gamut and range with BT.2020/2100. I mean heck, it wasn't that long ago that we barely had color at all. I still remember well the first 8-bit system I ever got, or back when I regularly had to manually change between 16-color/256/16k to prioritize resolution or color because my system just didn't have enough VRAM to handle both at once. Audio did far better at matching human limits much, much longer ago.
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.
Those are very nice examples. I always fell back on "go stare at the sun for a while for a shot of color that your monitor can't handle" but admittedly it was pure laziness.
Question: Is 192 kHz better when you want to slow down (or speed up) a track significantly while keeping pitches the same? Does it produce less noticeable artifacts?
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).
No, it's not. You answered your own question here:
> 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 personally read the article to be addressing 192 kHz as a consumer of the music, I have a feeling for those producing (or mixing, etc.) it's a bit different.
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.
Yeah, the author is not arguing against using 24 bits when recording, just when distributing to end users.
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.
I record and mix metal bands and have stuck with 48 kHz for years like many of the engineers I know. 96 kHz sounded better to my ears last time I checked in the studio (it's been years, maybe I wouldn't notice now that I'm older) but it's not worth the heavier storage and processing impact when nobody is actually going to use my stuff that way. I certainly don't feel limited by working at 48 kHz, either, but the hit to my workflow would be significant. Additionally, a lot of converters start imposing track limits when you go beyond 48 kHz, so that's one more reason to stay put.
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.
When slowing down the track, you're changing the effective sampling rate (e.g. 192 kHz turns into 96 kHz at half the speed). This article is about regular playback, so in your case it might make sense to have a higher rate.
Not likely. The DAC will almost invariably oversample anyway, and even though stretching may not happen in a high-sampling-frequency domain, the result eventually does.
The only rebuttal to this that I have found compelling is that 24/192 downloads make sense if you are going to sample the music in your own creations. Recording and mixing with extra dynamic range, combined with only needing to low-pass once at the end has demonstrable advantages. Of course this was a response to marketing that was definitely not targeted at samplers, so it's not so much of a rebuttal as arguing at cross points.
Yes, adding any sort of nonlinear distortion to audio will make frequencies depend on other frequencies, i.e. audible frequencies in the output of an effect can depend on supersonic frequencies in its input. For example, if you add a 100 kHz sine wave through a high-gain guitar amplifier, you'll definitely be able to hear it.
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.
He mentions this in the section "192kHz considered harmful" without the misleading rubbishing of Apple earbuds (which are among the best regular earbuds on the market, for what it's worth).
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.
Cymbals specifically produce a lot of ultrasonic audio, and some high sample rate recordings actually capture it. If you slow them down enough you can hear the difference.
When I casually researched the upper limit of human hearing, I came across something that mentioned that some people can detect lowpass filtering up to 27khz.
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.
I've been in an internet argument among very serious digital audio experts (such as from Bell Labs) where the consensus reached was this: for properly done audio export as a final stage to be heard by the most critical listeners, and by properly done I mean the output is dithered and not simply truncated and everything else is done properly:
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.
> When I casually researched the upper limit of human hearing, I came across something that mentioned that some people can detect lowpass filtering up to 27khz.
I'd want to know (a) if it was an analog or digital filter, (b) if the >20kHz signal intensity was normal/plausible and (c) how they ensured that the playback system wasn't generating intermodulation distortion products.
I'd be happy with CD quality - usually I have more than enough download bandwidth and storage space for it. Apple has had Apple Lossless for years but Apple Music (and the iTunes store) still use(s) lossy compression. Movies are now 4K, but Apple has been stuck on 256Kbps AAC since 2009. :(
Though as others have noted CD quality won't improve a terribly mastered recording from the loudness wars.
I wonder about one thing. Sure you can't hear above/under certain frequencies, but these frequencies still resonate with parts of your body (that are not ears) and you might feel it in other ways than just hearing, and also their presence generates harmonics. Not sure if it is observable to a human, but just because you don't hear N hertz, doesn't mean you can't hear its harmonics/it doesn't affect your _perception_ of the rest of the signal at all. Cutting off some frequencies can create patterns that are not hearable per se, but might induce unwanted sensory feelings (*opinion, not a fact). I think using physics to break this down doesn't make much sense, and that the most practical debate solution would be to do a double-blind test on a statistical group of the so called audiophiles.
I have a related question that someone here probably has a good answer for. I recently heard a song I like on the radio while driving. Shortly after it played I pulled up the same song on Spotify with my phone, plugged my phone into the car stereo through the headphone jack, and played it. The quality was MUCH worse. What's the likely reason for that?
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.
Radio stations often do additional processing of music to make it louder and more crisp when played on a car stereo, often by using techniques such as multi-band compression: the sounds is decomposed into several bands, and each band has dynamic range compression applied with different parameters to maximize the perceived sharpness/loudness.
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.
Spotify definitely sends you low quality audio at times. Most people won't notice on common speakers and headphones, but even my "bad" speakers revealed the difference.
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.
The other thing to note is that Spotify will send you lower quality audio on the mobile vs. desktop client.
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!
There is no "high definition radio." In the context of FM radio the H stands for hybrid and the D stands for digital. The digital often sounds worse when you compare them. Less noise for sure, but synthetic treble, almost as bad as Sirius XM.
Digital radio is very very low bitrate. A good FM signal is superior.
FM is capable of the same frequency response as CD, and it's a purely analog signal. If you don't have interference, you are getting a very pure stream of audio. The station is also probably using CD grade audio, so most of that quality is preserved.
From an audio standpoint, FM is a pretty decent sound medium. It's going to depend on your equipment, signal, and if you are moving.
There's quality setting in app settings, where you can choose the quality. Choices are automatic, low, normal, high very high. I guess "automatic" can adjust the quality based on connection.
I don't know if this is still common practice, but radio edits were often mastered differently back when I briefly studied music production. The station also likely uses signal processing to compress the dynamic range and increase the "loudness". Listening in a car is quite different than listening on home audio equipment, hence the different processing.
Spotify has compressed audio at pretty low bitrate. The FM signal has a full 20hz to 20khz frequency response and they are probably playing CD quality audio. Since it's just raw analog audio represented by radio frequency, it sounds better than the compressed stuff on spotify.
If you have a good signal, FM can be very high fidelity. Ever since people ditched CDs we've been listening to low quality streams and rips.
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.
> 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.
High resolution audio is important to me as a sound designer because of the ability to severely slow down a piece of audio without any aliasing or stuttering.
At 96KHz and higher with certain samples I can slow down by 80% and it will still sound good.
Do you mean slowing down while lowering pitch (without resampling)? If so, you're correct, as you bring harmonics from out of limit of human hearing back and the result sounds natural.
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.
Mastering should be done for studio monitors.
No, studio monitors do not sound the same, but they are somewhat neutral, they sound somewhat in the same ballpark, which is the point of them to begin with, have a flat frequency response (which does not imply that they "sound flat" just that music mastered for active-subwoofers sound flat).
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.
TL;DR: 24/192 is useful at the mastering stage, to avoid error creep from mixing and effects. This is way beyond human hearing however, so scaling down to 16/44.1 (CD quality) in the final mix for playback won't result in noticeable degradation. CD quality was chosen on principles of human limits.
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.
Like how banks keep track of fractional pennies when calculating interest, because rounding them off at time of calculation would introduce cumulative error. Instead, they round them off at payment time.
Storage space is cheap, and we have the ability to record and store music in 24/192 or any other format we want. Even if it's useless to us now, it may be of value some day when our genetically-engineered descendants can hear up to 40khz or when someone invents a direct condenser-microphone-to-brain interface.
Please do not confuse the usefulness of 24/192 for playback and listening enjoyment, and it's usefulness for recording and heavy 'in the box' processing.
In the box processing uses 32-bit or 64-bit float. Fixed-point DSP processing was a thing maybe ten years ago, and even then the standard was 56-bit. 24-bit is nowhere close to good enough for ITB DSP.
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.
Very inadequately. There was a quadrophonic vinyl system that failed commercially, which played back surround speakers using modulation of a 30K carrier tone. You had to use a special (aka 'good') stylus, and it sort of worked. The resulting carrier tones would go from 18 kHz to 45 kHz and the fact that this worked at all is evidence that vinyl goes up that far if you let it: wear will tend to scrub off that information unless it's a high energy transient, in which case there's a big chunk of plastic refusing to be worn off (but you'll dull it).
Even if the lower sample rate of 48kHz would be entirely reasonable and 96kHz is overkill, 24 bits still makes an audible difference for the material I listen to (modernist classical music and ECM jazz), which is why you can find 24/48 from some labels. For pop music, which of course is distinguished by little dynamic range, then 16 bits would be fine just like on the CD format.
I'm curious why you torrent music still when streaming is so widely available and free/cheap?
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.
I spend about half of every year traveling, often in particularly undeveloped countries and/or far from a mobile signal. Having my entire music collection on a portable hard drive is more convenient for me personally than being bound to streaming.
Streaming music services lack all the options in foobar2000 that I've grown accustomed to over the last 10+ years.
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.
>I'm curious why you torrent music still when streaming is so widely available and free/cheap?
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.
I refuse to pay streaming subs, I buy second hand CDs for pennies and rip to flac. I'll always own my content and play it whenever/wherever I want at the best quality.
Problem with music sounding bad doesn’t really have much to do with the distributed format: V0, V1, or 320 mp3s should sound pretty much the same compared to 16-bit flac. You can only the difference between mp3 and flac at shitty bitrates no one uses anymore (like 120).
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.
Since everyone is listening of crappy earbuds, they compress the hell out of it and destroy all dynamic range.
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.
> More compression and less dynamic range is beneficial for certain environments. Noisy subways. Watching TV in a noisy downtown apartment. Basically, crappy, noisy environments.
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.
But there are easy ways to kill dynamic range with an algorithm. On windows this is called "loudness equalization." On the otherhand, there is no way to go back from little dynamic range to more dynamic range.
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.
> 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.
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.
You can put brickwalled audio on a vinyl record and have it play just fine if you cut it at a lower volume. This negates the reason for mastering it that way in the first place, but it's cheaper than redoing the mastering, and many people buying vinyl only do so for the image. See:
NS10s are kicking around because they're unusually good at time domain performance. For instance, they have miserable bass not only because they're small boxes and smallish drivers, but because they're an infinite baffle design, which is significantly better for time domain performance than bass reflex. The enclosures also dissipate energy quite well, and it's well established that this contributes to being able to 'translate' mixes: you get a better sense of what's actually in the track using NS10s than you might with many 'better sounding' speakers.
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.
That may be true, but I've seen some vinyl mastering jobs that looked as bad as digital. I won't claim to be a mastering engineer or anything, but after comparing many vinyl releases and digital releases, it seems like there is something going on besides the physical limitations of the medium.
> modern vinyl releases sound a lot better than digital
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.
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.
Couldn’t agree more. I grew up with cassette and LP. First time I heard a CD, specifically Pink Floyd’s Money with the cash register, it was jaw dropping. LPs are cool for the artwork, but that’s it.
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.
"Since everyone is listening of crappy earbuds, they compress the hell out of it and destroy all dynamic range."
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 website is a useful resource, but it has some limitations. The algorithm used does not take into account the frequency response of the human ear. If a track contains a lot of very deep bass, it's possible for it to have a low DR score but still sound like it has a high dynamic range. The measurement can also be fooled by surface noise and filtering when measuring vinyl:
MP3s at any bitrate cannot properly reproduce certain sounds, as preecho can happen and it is fairly easy to train yourself to notice it. Pretty much any modern lossy format can be made transparent at high enough bitrates though.
My anecdotal experience is that music sounds the best to me currently via my good headphones connected to either my phone's good DAC, or to my PC's separate sound card.
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.
Vinyl releases are mastered to be less loud than digital releases because vinyl cannot reproduce mixes that digital systems can. The side effect is that lots of times they sound better. I think in a perfect world an artist would offer you vinyl if you want it, along with a digital version of the vinyl master. You could skip the whole ripping vinyl process entirely.
One of the "nice" things about being hard of hearing is that I can't hear any difference between flac and mp3s down to around 96 or lower for most music, so hypothetically I don't have to worry about this stuff.
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.
So vinyl has only about the equivalent of 10-14 bits of resolution (I don't remember the exact number I heard and it has been a while) and waveforms within our hearing range are far larger than what 192khz can potentially accomodate. The only use I've found for such high-resolution is audio is using it as base material for further effects processing... certain distortion units and whatnot that operate on a sample level can sometimes give nicer output when fed super hi-res audio
No, not at all. Vinyl has a wildly inconsistent noise level where rumble predominates, and people conflate this with bits of resolution. Vinyl's behavior is not easily pinned down relative to 'bits of resolution', because the noise floor is skewed so intensely towards low frequencies.
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.
Yes, but we are not talking about the mixing/mastering sample rate, but the distribution sample rate/resolution.
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.
> 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.
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...
The problem is that they aren't being mastered differently - there's a website that lists vinyl releases (can't find the link) and compares it to the CD masters and they're often the same thing. Older CD masters from the 80's or 90's are re-released compressed to drive sales. The latest vinyl fad has just become a new means for record companies to to exploit a "new" medium and race for the bottom - they know many new listeners on cheap players actually just want to hear what they get out of the earbud.
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.
Some time ago, I stumbled across a YT channel of some guy, a self-professed studio expert, who "remasters" some 80's metal albums to give them a big, "modern" sound. The uploads are heavily commented with positive reviews.
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.
Back up for a second with the last paragraph there: if the record was mastered for a room sized stereo, it assumes that the room adds its reverb to the sound. With loudspeakers, the room is a distorting filter in the signal path. This and the HRTF distortion are skipped over when listening to headphones/earbuds. So it does make a lot of sense to add these effects to the audio signal in the headphones case. Done right, the headphone playback is indistinguishable from a stereo in a room - mounted to your head, because the spatialized speakers are relative to your head, no matter where you look.
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.
Exactly if it’s mastered wrong the nitrate has nothing to do with the issue this music would sound just as awful cut to vinyl from a bad master. Now you just can’t hear beyond 22050 so 192 is insanely wasteful. But poor mastering is absolutely the core issue not encoding algorithms
Actually, no, it might sound better cut to vinyl. Remember, vinyl doesn't have the frequency range or dynamic range that digital audio does, and it has to be mastered using the RIAA Curve because of the properties of the medium. One factor here is that the stereo separation on vinyl can't be too large, or else the needle will literally jump out of the groove! In short, you can't just take CD music (no matter how well or poorly mastered) and cut it to vinyl as-is.
> 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.
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 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 hike a lot and hate using my phone's battery power for music. On top of needing that power for other things, it just feels wasteful. I bought a cheap MP3 player to try out in 2016 and have been hooked ever since. These devices are smaller and lighter than spare phone batteries or power banks.
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.
But aren't you worried you'll lose access to your music? I have to own it! I can't have it at the whim of multiple third parties to take down as they see fit. It's too important.
> I'm not sure but I think "owning" music like in "I got some files here on my drive" seems dead to me.
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.
> But who is using MP3 players these days any more?
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.
I rip my CDs in a two-step process: first to FLAC, then convert to mp3. The mp3s go in my phone, I have 33GB so far and my collection isn't even half ripped. I haven't checked how big the FLACs are lately but I'm sure they'd be a much bigger burden.
Well I never used CDs. Unfortunately what.cd got taken down, but a couple years ago, it was probably the biggest and most complete collection of music in the world.
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.
I use my phone as an MP3 (Opus, actually) player, with a selection of music from my ~20K track collection. This works better for me than unlimited access to all music, because it makes me have to listen to a smaller selection of content, so I give each album more attention.
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.
Your smartphone or laptop is like an MP3 player with respect to mastering, not like an expensive amplifier and speakers. Your smaryphone/laptop has an amplifier that's optimised for low energy usage, not fidelity, and loudspeakers optimised for size. Music which has been mixed and mastered without regard for how it sounds on your smartphone is sold as "24/192" or "vinyl" or such. The 192 does not matter technically, it's just an identifying mark, and some sort of identifying mark is necessary.
> 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.
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
Oh another cool thing about vinyl is the needle can couple to the environment too, try driving its case with another speaker or putting it in front of it's own big amp for feedback.
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.
> 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.
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
> That being said, I think flac is generally a good choice for a music collection.
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.
Most LPs these days are made from the same masters as the CDs (or downloads/streaming), with only the bare minimum of processing done to make them viable to pressing to vinyl, ie. mono bass and RIAA equalization. Only releases marketed specifically to audiophiles tend to get any extra effort put into them, and that is a vanishingly small segment of customers.
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 2019 music streaming should be more like video streaming, in that different bitrates should be user selectable, and processing ("cinema" vs. "night mode") is done by the playback equipment.
> mp3s should sound pretty much the same compared to 16-bit flac
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.
There is some definite perceptible loss in accuracy in the treble even at V0 or 320. There's a song with a synthesized treble effect that sounds quite different on MP3 vs. FLAC by Planet Funk, I think it was "Who Said? (Stuck in the UK)".
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.
Interesting. Are you sure you were using the right settings and a recent version of LAME for this?
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.
You also lose some of the fullness on the extreme low end, it's noticeable even with a fairly low end subwoofer.
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.
It did fail, but note that it did sound freaking awesome. I have one. The thing rivals all the rest of my studio gear which includes a Lavry Black and a MOTU 16A capable of 24/192k (which doesn't actually sound better than the Black, but does spec better)
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!)
This is a good example of the Dunning–Kruger effect. Guy reads some books on a subject and thinks he understands all there is about it and think everybody else is stupid.
I'm not a big believer of audiophile stuff, but when I'm listening to 24k music I'm hearing new instruments, new sounds. It's not the case with everything though. Am I retarded ?
author doesn't take into account that although you cannot hear above 20khz or below 20hz. That doesn't mean you cannot sense it. Sound after all is just air vibrating, therefore obviously there must be an effect on the body.
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.
And like homeopathy, this should be scrutinized scientifically. There is nothing wrong with eating tiny bits of sugared balls, but don't tell others they are somehow of special powers.
I have a degree in electrical engineering, and I'm currently in a graduate course on computer music systems, so I hope that qualifies me enough to avoid the author's ad hominem attacks he seasons this stinkpiece with.
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 have a hard-won TDS meter that lets me measure the total dissolved solids in a cup of coffee.
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.
So you think our senses transcend what tools have the ability to measure? Maybe that was the case in 1970, but in 2000+, hearing (and vision) is completely understood scientifically and far surpassed by measuring apparatus at every frequency range. Saying otherwise is an appeal to what is called audio mysticism and is caused by placebo and confirmation bias, which was mentioned in the article.
> [I]n 2000+, hearing (and vision) is completely understood scientifically and far surpassed by measuring apparatus at every frequency range
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.
Actually that's a defensible position when you're mixing and mastering audio. A spectrogram won't tell you which settings sound better, but your ears will.
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.
If it was true, the entirety of hi-fi industry would not exist. I myself built a number of amplifiers and, after a certain threshold, roughly .02% THD (total harmonic distortion) at 20khz, there is a very little correlation between the THD (what is usually measured)numbers and perceived quality of the sound. Which means, while it is true that is everything could be measured, no one measures the right thing (perhaps some weird subtle phase shifts or some almost immeasurable frequency response deficiencies)
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.
Listening to 16bit 44.1 kHz can also damage the cilia in your ears. Whether it actually does, in both cases, of course depends on the recorded material and the playback gain.
Bingo! Exactly, Nyquist theory. But no one can make a truly good, brickwall filter at this frequency. It is much easier to make a lot less steep frequency response, whilch would start at 22 khz and would slowly go down all the way to 96 kHz.
By the way, how would it damage the ears if it was thoroughly filtered?
> "Health 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 medicine."
Would you accept that argument from someone purporting to provide medical advice?
Already discussed. Main posts:
[2012] https://news.ycombinator.com/item?id=15127633
It's a very good article which shows up again and again. 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 .
It is a good article, and since the misunderstandings are persistent in the same way a lot of other commercially exploited mysticism is it remains a relevant one as well. Having said that and to try to add on something new to these discussions, since you brought up this:
>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.
1 reply →
Could it be also because HN Front page algorithm (my speculation) of favoring domains (with something like Domain Authority) based on previous votes?
1 reply →
Feels like we should have spaced repetition/Anki for highly-upvoted articles
It’s supposed to have (2012) in the title if it’s an old article.
This reminds me of Monty's A Digital Media Primer for Geeks[0] and Digital Show & Tell[1] - the delivery, the explanations and the way the experiments are set up is superb.
[0] https://xiph.org/video/vid1.shtml [1] https://xiph.org/video/vid2.shtml
The article's author, Chris "Monty" Montgomery, is one of the authors of Ogg Vorbis [1] and Opus [2].
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, ...)."
[1] https://xiph.org/vorbis/
[2] http://www.opus-codec.org/comparison/
[3] https://wiki.xiph.org/OpusFAQ#Does_Opus_make_all_those_other...
I use Opus for music playback for all my archived music. The reason it's not more widespread was opposition of the likes of Apple to free codecs. Today they are losing this, and Opus is making its way even to Apple's systems.
2 replies →
I love opus just as much as I loved musepack and vorbis, the one thing all of them lack to one degree or another is support and hardware acceleration. If I throw an opus file on my Android 8.1 phone, it has no idea what to do with it unless I manually open it with vlc or foobar. For the regular user the support needs to be seamless, otherwise they are not going to bother.
I thought for a moment that Spotify uses Opus, but it turns out that they use Vorbis. Wonder why a switch isn't on their roadmap.
4 replies →
The use of analogue gear in #2 is one of those things that as someone who _already believed what Monty is showing here_ I wouldn't have thought to do. But it really heads off a bunch of arguments.
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.
Honestly it is remarkable how many engineers (self-proclaimed or otherwise) in audio don't understand the basics of sampled systems and quantization. You'd think that anyone making broad claims about these kinds of systems would have at least a rough understanding of the foundational principles, but no.
The choice of colour analogy is unfortunate, because there really are colours that are "out of gamut" and cannot be accurately reproduced on normal monitors. If you have the opportunity to and look at one of the IKB works in person you'll see what I mean.
https://www.tate.org.uk/art/artworks/klein-ikb-79-t01513
https://contemporaryartetc.wordpress.com/2007/09/13/fact-of-...
I don't quite agree with you, taking into account it's an analogy designed to help illustrate the issue for general audiences. It's not as if we don't have ProPhoto RGB or other wide gamuts or don't understand the issues of rendition accuracy and resolution within the visual spectrum. There was never any debate that sRGB alone in particular was quite limited, or that dynamic range was an issue. It's just that it represents a ton more data and is technologically and commercially much, much harder. As tech has caught up displays have continued to chase human visual limits, starting with resolution, then frame rate, and finally major industry wide improvements to gamut and range with BT.2020/2100. I mean heck, it wasn't that long ago that we barely had color at all. I still remember well the first 8-bit system I ever got, or back when I regularly had to manually change between 16-color/256/16k to prioritize resolution or color because my system just didn't have enough VRAM to handle both at once. Audio did far better at matching human limits much, much longer ago.
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.
Those are very nice examples. I always fell back on "go stare at the sun for a while for a shot of color that your monitor can't handle" but admittedly it was pure laziness.
Question: Is 192 kHz better when you want to slow down (or speed up) a track significantly while keeping pitches the same? Does it produce less noticeable artifacts?
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).
No, it's not. You answered your own question here:
> 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 personally read the article to be addressing 192 kHz as a consumer of the music, I have a feeling for those producing (or mixing, etc.) it's a bit different.
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.
Yeah, the author is not arguing against using 24 bits when recording, just when distributing to end users.
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.
1 reply →
I record and mix metal bands and have stuck with 48 kHz for years like many of the engineers I know. 96 kHz sounded better to my ears last time I checked in the studio (it's been years, maybe I wouldn't notice now that I'm older) but it's not worth the heavier storage and processing impact when nobody is actually going to use my stuff that way. I certainly don't feel limited by working at 48 kHz, either, but the hit to my workflow would be significant. Additionally, a lot of converters start imposing track limits when you go beyond 48 kHz, so that's one more reason to stay put.
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.
When slowing down the track, you're changing the effective sampling rate (e.g. 192 kHz turns into 96 kHz at half the speed). This article is about regular playback, so in your case it might make sense to have a higher rate.
Not likely. The DAC will almost invariably oversample anyway, and even though stretching may not happen in a high-sampling-frequency domain, the result eventually does.
The only rebuttal to this that I have found compelling is that 24/192 downloads make sense if you are going to sample the music in your own creations. Recording and mixing with extra dynamic range, combined with only needing to low-pass once at the end has demonstrable advantages. Of course this was a response to marketing that was definitely not targeted at samplers, so it's not so much of a rebuttal as arguing at cross points.
Yes, adding any sort of nonlinear distortion to audio will make frequencies depend on other frequencies, i.e. audible frequencies in the output of an effect can depend on supersonic frequencies in its input. For example, if you add a 100 kHz sine wave through a high-gain guitar amplifier, you'll definitely be able to hear it.
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.
He mentions this in the section "192kHz considered harmful" without the misleading rubbishing of Apple earbuds (which are among the best regular earbuds on the market, for what it's worth).
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.
Cymbals specifically produce a lot of ultrasonic audio, and some high sample rate recordings actually capture it. If you slow them down enough you can hear the difference.
yup; from the piece: "An engineer also requires more than 16 bits during mixing and mastering"
It's also not so likely that a given recording would actually have the larger dynamic range.
When I casually researched the upper limit of human hearing, I came across something that mentioned that some people can detect lowpass filtering up to 27khz.
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.
I've been in an internet argument among very serious digital audio experts (such as from Bell Labs) where the consensus reached was this: for properly done audio export as a final stage to be heard by the most critical listeners, and by properly done I mean the output is dithered and not simply truncated and everything else is done properly:
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.
> When I casually researched the upper limit of human hearing, I came across something that mentioned that some people can detect lowpass filtering up to 27khz.
Link?
I'd want to know (a) if it was an analog or digital filter, (b) if the >20kHz signal intensity was normal/plausible and (c) how they ensured that the playback system wasn't generating intermodulation distortion products.
https://en.wikipedia.org/wiki/Hearing_range#Humans
1 reply →
If you use foobar2000, you can use ABX Comparator to compare between various bitrates and formats. Start with a lossless format and convert it.
[1] https://www.foobar2000.org/components/view/foo_abx
I'd be happy with CD quality - usually I have more than enough download bandwidth and storage space for it. Apple has had Apple Lossless for years but Apple Music (and the iTunes store) still use(s) lossy compression. Movies are now 4K, but Apple has been stuck on 256Kbps AAC since 2009. :(
Though as others have noted CD quality won't improve a terribly mastered recording from the loudness wars.
I wonder about one thing. Sure you can't hear above/under certain frequencies, but these frequencies still resonate with parts of your body (that are not ears) and you might feel it in other ways than just hearing, and also their presence generates harmonics. Not sure if it is observable to a human, but just because you don't hear N hertz, doesn't mean you can't hear its harmonics/it doesn't affect your _perception_ of the rest of the signal at all. Cutting off some frequencies can create patterns that are not hearable per se, but might induce unwanted sensory feelings (*opinion, not a fact). I think using physics to break this down doesn't make much sense, and that the most practical debate solution would be to do a double-blind test on a statistical group of the so called audiophiles.
I have a related question that someone here probably has a good answer for. I recently heard a song I like on the radio while driving. Shortly after it played I pulled up the same song on Spotify with my phone, plugged my phone into the car stereo through the headphone jack, and played it. The quality was MUCH worse. What's the likely reason for that?
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.
Radio stations often do additional processing of music to make it louder and more crisp when played on a car stereo, often by using techniques such as multi-band compression: the sounds is decomposed into several bands, and each band has dynamic range compression applied with different parameters to maximize the perceived sharpness/loudness.
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."
Spotify definitely sends you low quality audio at times. Most people won't notice on common speakers and headphones, but even my "bad" speakers revealed the difference.
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.
The other thing to note is that Spotify will send you lower quality audio on the mobile vs. desktop client.
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!
3. The A/D converter used by your car stereo headphone jack is low quality and introduced sampling artifacts.
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.
There is no "high definition radio." In the context of FM radio the H stands for hybrid and the D stands for digital. The digital often sounds worse when you compare them. Less noise for sure, but synthetic treble, almost as bad as Sirius XM.
Digital radio is very very low bitrate. A good FM signal is superior.
FM is capable of the same frequency response as CD, and it's a purely analog signal. If you don't have interference, you are getting a very pure stream of audio. The station is also probably using CD grade audio, so most of that quality is preserved.
From an audio standpoint, FM is a pretty decent sound medium. It's going to depend on your equipment, signal, and if you are moving.
There's quality setting in app settings, where you can choose the quality. Choices are automatic, low, normal, high very high. I guess "automatic" can adjust the quality based on connection.
I don't know if this is still common practice, but radio edits were often mastered differently back when I briefly studied music production. The station also likely uses signal processing to compress the dynamic range and increase the "loudness". Listening in a car is quite different than listening on home audio equipment, hence the different processing.
Spotify has compressed audio at pretty low bitrate. The FM signal has a full 20hz to 20khz frequency response and they are probably playing CD quality audio. Since it's just raw analog audio represented by radio frequency, it sounds better than the compressed stuff on spotify.
If you have a good signal, FM can be very high fidelity. Ever since people ditched CDs we've been listening to low quality streams and rips.
This same argument can be made for many things
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.
Your examples all have good reasons though.
> 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.
More analogies—
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.
1 reply →
> 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.
High resolution audio is important to me as a sound designer because of the ability to severely slow down a piece of audio without any aliasing or stuttering.
At 96KHz and higher with certain samples I can slow down by 80% and it will still sound good.
Do you mean slowing down while lowering pitch (without resampling)? If so, you're correct, as you bring harmonics from out of limit of human hearing back and the result sounds natural.
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.
Mastering should be done for studio monitors. No, studio monitors do not sound the same, but they are somewhat neutral, they sound somewhat in the same ballpark, which is the point of them to begin with, have a flat frequency response (which does not imply that they "sound flat" just that music mastered for active-subwoofers sound flat).
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.
I store stuff in flac, i got a large nas at my house. Then i can down convert to any other format I might need. I enjoy the flacs when I'm home.
TL;DR: 24/192 is useful at the mastering stage, to avoid error creep from mixing and effects. This is way beyond human hearing however, so scaling down to 16/44.1 (CD quality) in the final mix for playback won't result in noticeable degradation. CD quality was chosen on principles of human limits.
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.
Like how banks keep track of fractional pennies when calculating interest, because rounding them off at time of calculation would introduce cumulative error. Instead, they round them off at payment time.
Yes, as done in an image compositing pipelines as well, with higher resolutions and bits per pixel of color information until final output.
Storage space is cheap, and we have the ability to record and store music in 24/192 or any other format we want. Even if it's useless to us now, it may be of value some day when our genetically-engineered descendants can hear up to 40khz or when someone invents a direct condenser-microphone-to-brain interface.
Please do not confuse the usefulness of 24/192 for playback and listening enjoyment, and it's usefulness for recording and heavy 'in the box' processing.
In the box processing uses 32-bit or 64-bit float. Fixed-point DSP processing was a thing maybe ten years ago, and even then the standard was 56-bit. 24-bit is nowhere close to good enough for ITB DSP.
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.
Very inadequately. There was a quadrophonic vinyl system that failed commercially, which played back surround speakers using modulation of a 30K carrier tone. You had to use a special (aka 'good') stylus, and it sort of worked. The resulting carrier tones would go from 18 kHz to 45 kHz and the fact that this worked at all is evidence that vinyl goes up that far if you let it: wear will tend to scrub off that information unless it's a high energy transient, in which case there's a big chunk of plastic refusing to be worn off (but you'll dull it).
> That aside - the bit depth part of this article is silly and wrong. With an unprocessed acoustic recording
It's neither silly nor wrong—the article's title literally excludes it from consideration. This is about music downloads, not music production.
My favorite music these days is left-field and lo-fi hip hop. High-fidelity is pretty irrelevant. Music is produced, mixed, and mastered at home.
e.g. https://pitchfork.com/reviews/albums/earl-sweatshirt-some-ra...
Thanks to this article, when I torrent music that is shared in 24/192 format, I always resample to 24/96:
ffmpeg -i foo.flac -ar 96000 -acodec flac bar.flac
To thank the article even more, you could resample to 16/48kHz.
Even if the lower sample rate of 48kHz would be entirely reasonable and 96kHz is overkill, 24 bits still makes an audible difference for the material I listen to (modernist classical music and ECM jazz), which is why you can find 24/48 from some labels. For pop music, which of course is distinguished by little dynamic range, then 16 bits would be fine just like on the CD format.
8 replies →
I'm curious why you torrent music still when streaming is so widely available and free/cheap?
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.
I spend about half of every year traveling, often in particularly undeveloped countries and/or far from a mobile signal. Having my entire music collection on a portable hard drive is more convenient for me personally than being bound to streaming.
1 reply →
Streaming music services lack all the options in foobar2000 that I've grown accustomed to over the last 10+ years.
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.
>I'm curious why you torrent music still when streaming is so widely available and free/cheap?
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.
I refuse to pay streaming subs, I buy second hand CDs for pennies and rip to flac. I'll always own my content and play it whenever/wherever I want at the best quality.
Some bands still refuse to be available on streaming (ex: Tool). Some will never be on streaming.
Problem with music sounding bad doesn’t really have much to do with the distributed format: V0, V1, or 320 mp3s should sound pretty much the same compared to 16-bit flac. You can only the difference between mp3 and flac at shitty bitrates no one uses anymore (like 120).
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.
Since everyone is listening of crappy earbuds, they compress the hell out of it and destroy all dynamic range.
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.
> More compression and less dynamic range is beneficial for certain environments. Noisy subways. Watching TV in a noisy downtown apartment. Basically, crappy, noisy environments.
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.
2 replies →
But there are easy ways to kill dynamic range with an algorithm. On windows this is called "loudness equalization." On the otherhand, there is no way to go back from little dynamic range to more dynamic range.
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.
4 replies →
Dynamic range is a solved problem if people cared.
You attach some metadata to the audio file that says certain parts should be level boosted in a noisy environment and there you go.
4 replies →
In the 1990s, car stereos sometimes had a "loudness" button which did exactly as you suggest.
4 replies →
Isn’t radio a big factor in this? Broadcast radio is noisy and has pretty limited dynamic range. This may be a cause.
5 replies →
> then appropriate signal processing should be done by playback.
Microchips for leveling audio gain existed in the 1980's and were found in consumer equpiment like TV's.
> 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.
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.
You can put brickwalled audio on a vinyl record and have it play just fine if you cut it at a lower volume. This negates the reason for mastering it that way in the first place, but it's cheaper than redoing the mastering, and many people buying vinyl only do so for the image. See:
https://wiki.hydrogenaud.io/index.php?title=Myths_(Vinyl)#My...
1 reply →
NS10s are kicking around because they're unusually good at time domain performance. For instance, they have miserable bass not only because they're small boxes and smallish drivers, but because they're an infinite baffle design, which is significantly better for time domain performance than bass reflex. The enclosures also dissipate energy quite well, and it's well established that this contributes to being able to 'translate' mixes: you get a better sense of what's actually in the track using NS10s than you might with many 'better sounding' speakers.
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.
That may be true, but I've seen some vinyl mastering jobs that looked as bad as digital. I won't claim to be a mastering engineer or anything, but after comparing many vinyl releases and digital releases, it seems like there is something going on besides the physical limitations of the medium.
> modern vinyl releases sound a lot better than digital
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.
Couldn’t agree more. I grew up with cassette and LP. First time I heard a CD, specifically Pink Floyd’s Money with the cash register, it was jaw dropping. LPs are cool for the artwork, but that’s it.
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.
2 replies →
"Since everyone is listening of crappy earbuds, they compress the hell out of it and destroy all dynamic range."
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...
Here is a link[0] to CD dynamic range database from where you can check how a particular mastering fares.
[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.
This website is a useful resource, but it has some limitations. The algorithm used does not take into account the frequency response of the human ear. If a track contains a lot of very deep bass, it's possible for it to have a low DR score but still sound like it has a high dynamic range. The measurement can also be fooled by surface noise and filtering when measuring vinyl:
https://wiki.hydrogenaud.io/index.php?title=Myths_(Vinyl)#Ef...
MP3s at any bitrate cannot properly reproduce certain sounds, as preecho can happen and it is fairly easy to train yourself to notice it. Pretty much any modern lossy format can be made transparent at high enough bitrates though.
Example?
9 replies →
My anecdotal experience is that music sounds the best to me currently via my good headphones connected to either my phone's good DAC, or to my PC's separate sound card.
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.
Vinyl releases are mastered to be less loud than digital releases because vinyl cannot reproduce mixes that digital systems can. The side effect is that lots of times they sound better. I think in a perfect world an artist would offer you vinyl if you want it, along with a digital version of the vinyl master. You could skip the whole ripping vinyl process entirely.
One of the "nice" things about being hard of hearing is that I can't hear any difference between flac and mp3s down to around 96 or lower for most music, so hypothetically I don't have to worry about this stuff.
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.
So vinyl has only about the equivalent of 10-14 bits of resolution (I don't remember the exact number I heard and it has been a while) and waveforms within our hearing range are far larger than what 192khz can potentially accomodate. The only use I've found for such high-resolution is audio is using it as base material for further effects processing... certain distortion units and whatnot that operate on a sample level can sometimes give nicer output when fed super hi-res audio
No, not at all. Vinyl has a wildly inconsistent noise level where rumble predominates, and people conflate this with bits of resolution. Vinyl's behavior is not easily pinned down relative to 'bits of resolution', because the noise floor is skewed so intensely towards low frequencies.
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.
3 replies →
Yes, but we are not talking about the mixing/mastering sample rate, but the distribution sample rate/resolution.
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.
> 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.
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...
At minimum they have to move all the bass to the center and apply RIAA EQ.
You're certainly right, many aren't (I always check). A lot are though. I'd give it maybe like 50/50.
The problem is that they aren't being mastered differently - there's a website that lists vinyl releases (can't find the link) and compares it to the CD masters and they're often the same thing. Older CD masters from the 80's or 90's are re-released compressed to drive sales. The latest vinyl fad has just become a new means for record companies to to exploit a "new" medium and race for the bottom - they know many new listeners on cheap players actually just want to hear what they get out of the earbud.
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.
Some time ago, I stumbled across a YT channel of some guy, a self-professed studio expert, who "remasters" some 80's metal albums to give them a big, "modern" sound. The uploads are heavily commented with positive reviews.
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.
Back up for a second with the last paragraph there: if the record was mastered for a room sized stereo, it assumes that the room adds its reverb to the sound. With loudspeakers, the room is a distorting filter in the signal path. This and the HRTF distortion are skipped over when listening to headphones/earbuds. So it does make a lot of sense to add these effects to the audio signal in the headphones case. Done right, the headphone playback is indistinguishable from a stereo in a room - mounted to your head, because the spatialized speakers are relative to your head, no matter where you look.
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.
2 replies →
Exactly if it’s mastered wrong the nitrate has nothing to do with the issue this music would sound just as awful cut to vinyl from a bad master. Now you just can’t hear beyond 22050 so 192 is insanely wasteful. But poor mastering is absolutely the core issue not encoding algorithms
Actually, no, it might sound better cut to vinyl. Remember, vinyl doesn't have the frequency range or dynamic range that digital audio does, and it has to be mastered using the RIAA Curve because of the properties of the medium. One factor here is that the stereo separation on vinyl can't be too large, or else the needle will literally jump out of the groove! In short, you can't just take CD music (no matter how well or poorly mastered) and cut it to vinyl as-is.
3 replies →
> 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.
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.
But who is using MP3 players these days any more?
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 hike a lot and hate using my phone's battery power for music. On top of needing that power for other things, it just feels wasteful. I bought a cheap MP3 player to try out in 2016 and have been hooked ever since. These devices are smaller and lighter than spare phone batteries or power banks.
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.
But aren't you worried you'll lose access to your music? I have to own it! I can't have it at the whim of multiple third parties to take down as they see fit. It's too important.
19 replies →
> I'm not sure but I think "owning" music like in "I got some files here on my drive" seems dead to me.
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.
> But who is using MP3 players these days any more?
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.
I rip my CDs in a two-step process: first to FLAC, then convert to mp3. The mp3s go in my phone, I have 33GB so far and my collection isn't even half ripped. I haven't checked how big the FLACs are lately but I'm sure they'd be a much bigger burden.
31 replies →
Well I never used CDs. Unfortunately what.cd got taken down, but a couple years ago, it was probably the biggest and most complete collection of music in the world.
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.
2 replies →
I use my phone as an MP3 (Opus, actually) player, with a selection of music from my ~20K track collection. This works better for me than unlimited access to all music, because it makes me have to listen to a smaller selection of content, so I give each album more attention.
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.
Your smartphone or laptop is like an MP3 player with respect to mastering, not like an expensive amplifier and speakers. Your smaryphone/laptop has an amplifier that's optimised for low energy usage, not fidelity, and loudspeakers optimised for size. Music which has been mixed and mastered without regard for how it sounds on your smartphone is sold as "24/192" or "vinyl" or such. The 192 does not matter technically, it's just an identifying mark, and some sort of identifying mark is necessary.
I don't think this advice is aimed at your typical Spotify user (i.e. the majority of people).
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.
> 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.
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
Oh another cool thing about vinyl is the needle can couple to the environment too, try driving its case with another speaker or putting it in front of it's own big amp for feedback.
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.
> 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.
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
(Assuming you're okay with piracy…) You're better off searching for vinyl rips where people with good equipment have done the heavy lifting for you.
1 reply →
> That being said, I think flac is generally a good choice for a music collection.
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.
IIRC, XLD rips to WAV first anyway, then compresses it to FLAC (I know EAC does).
The reason why a lot of recent digital music sounds bad is because of the intentionally terrible mastering.
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.
Most LPs these days are made from the same masters as the CDs (or downloads/streaming), with only the bare minimum of processing done to make them viable to pressing to vinyl, ie. mono bass and RIAA equalization. Only releases marketed specifically to audiophiles tend to get any extra effort put into them, and that is a vanishingly small segment of customers.
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.
> (there are some exceptions, like the Beatles mono and stereo boxed sets that came out awhile ago)
Didn't the beatles famously create their music to be listenable on the terrible radios of the time?
So this is why streaming Google Music on my fairly nice sound system ends up sounding like total crap?
In 2019 music streaming should be more like video streaming, in that different bitrates should be user selectable, and processing ("cinema" vs. "night mode") is done by the playback equipment.
> mp3s should sound pretty much the same compared to 16-bit flac
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.
There is some definite perceptible loss in accuracy in the treble even at V0 or 320. There's a song with a synthesized treble effect that sounds quite different on MP3 vs. FLAC by Planet Funk, I think it was "Who Said? (Stuck in the UK)".
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.
Interesting. Are you sure you were using the right settings and a recent version of LAME for this?
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.
9 replies →
You also lose some of the fullness on the extreme low end, it's noticeable even with a fairly low end subwoofer.
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.
3 replies →
Should this have a (2012) tag? Neil Young did end up releasing music in this format with his Pono player, which failed[0].
[0]: http://www.noise11.com/news/r-i-p-pono-neil-young-kills-off-...
Oh, golly, that hot yellow mess. I bought one and it's still in a desk drawer somewhere. The audio sounded great, but the device itself was rather dire. Here's my notes on it at the time: https://tenfourfox.blogspot.com/2015/01/and-now-for-somethin...
It did fail, but note that it did sound freaking awesome. I have one. The thing rivals all the rest of my studio gear which includes a Lavry Black and a MOTU 16A capable of 24/192k (which doesn't actually sound better than the Black, but does spec better)
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!)
Yes, this has been their position for many years. The associated video is good too: https://xiph.org/video/vid2.shtml
Footer of the page suggests 2024, not 2012, as the latest edit date. You can ask the mods to add it by emailing them (link in HN footer).
2014, sigh.
Title should have [2012]
This is a good example of the Dunning–Kruger effect. Guy reads some books on a subject and thinks he understands all there is about it and think everybody else is stupid.
I'm not a big believer of audiophile stuff, but when I'm listening to 24k music I'm hearing new instruments, new sounds. It's not the case with everything though. Am I retarded ?
Confirmation bias. If you listened to the same track, but at 44.1/16, and were told it was hi-res, you would have the same reaction.
author doesn't take into account that although you cannot hear above 20khz or below 20hz. That doesn't mean you cannot sense it. Sound after all is just air vibrating, therefore obviously there must be an effect on the body.
Just to be a Devil's advocate (just a little)...
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.
And like homeopathy, this should be scrutinized scientifically. There is nothing wrong with eating tiny bits of sugared balls, but don't tell others they are somehow of special powers.
I have a degree in electrical engineering, and I'm currently in a graduate course on computer music systems, so I hope that qualifies me enough to avoid the author's ad hominem attacks he seasons this stinkpiece with.
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?
[1] https://www.youtube.com/watch?v=RlCG2fK-abo
I have a hard-won TDS meter that lets me measure the total dissolved solids in a cup of coffee.
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.
So you think our senses transcend what tools have the ability to measure? Maybe that was the case in 1970, but in 2000+, hearing (and vision) is completely understood scientifically and far surpassed by measuring apparatus at every frequency range. Saying otherwise is an appeal to what is called audio mysticism and is caused by placebo and confirmation bias, which was mentioned in the article.
> [I]n 2000+, hearing (and vision) is completely understood scientifically and far surpassed by measuring apparatus at every frequency range
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.
Actually that's a defensible position when you're mixing and mastering audio. A spectrogram won't tell you which settings sound better, but your ears will.
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.
3 replies →
Yes, that's what I think. I don't believe that hearing is completely understood scientifically. You're welcome to disagree.
3 replies →
If it was true, the entirety of hi-fi industry would not exist. I myself built a number of amplifiers and, after a certain threshold, roughly .02% THD (total harmonic distortion) at 20khz, there is a very little correlation between the THD (what is usually measured)numbers and perceived quality of the sound. Which means, while it is true that is everything could be measured, no one measures the right thing (perhaps some weird subtle phase shifts or some almost immeasurable frequency response deficiencies)
8 replies →
Sorry, I read your comment twice... do you actually dispute any of the points made in the article, or are you simply wishing that people didn't care?
I'll just respond by quoting the article:
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.
This was basically a PSA back when Neil Young was pushing the Pono music player and store. https://en.wikipedia.org/wiki/Pono_(digital_music_service)
Tidal still sells on this basis too.
If you really had any qualifications, that anti-scientific rant made the irrelevant.
Was this comment machine-generated? Every comment by your account is negative and unsubstantial.
> not "unexpected 14.5kHz falloff"
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.
If two wines were chemically identical but someone claimed that one of them tasted “warmer” than the other, would you just go with that?
Sure. I don't need to invalidate someone else's experience.
5 replies →
The is not the only scenario. How about you made some analysis of two wines and they seem to be same, yet someone can _reliably_ tell them apart?
3 replies →
There is the law of diminishing returns on audio quality, as in everything. Empirical data collection is not a thing to be avoided however.
Given that, your rant seems misplaced, and I can't help but think engineering is not the field for you.
Umm, Nyquist-Shannon Theory? Listening to 24Bit 192Khz can actually damage the cillia in your ear
Listening to 16bit 44.1 kHz can also damage the cilia in your ears. Whether it actually does, in both cases, of course depends on the recorded material and the playback gain.
Bingo! Exactly, Nyquist theory. But no one can make a truly good, brickwall filter at this frequency. It is much easier to make a lot less steep frequency response, whilch would start at 22 khz and would slowly go down all the way to 96 kHz. By the way, how would it damage the ears if it was thoroughly filtered?
8 replies →
> "Health 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 medicine."
Would you accept that argument from someone purporting to provide medical advice?