It’s interesting that the author seems keen to discuss the concept of psychoacoustic encoding only to dedicate much of the article to looking at waveforms and making condemnations based on looking at those waveforms. Which misses the point entirely.
This is a somewhat older article, but most lossy encoders at the time were still very good even at fairly low (sub-96 kbps) bit rates.
The author explains that it's a psychoacoustic model and then demonstrates how various artifacts in various waveforms are the result of compromises within this model. As an aside, I don't understand why people take the position that lossy transcodes are indistinguishable from one another when, right out of the gate, it's well-understand that compromises are being made in the highs, well within the range of human hearing. In fact, CDs can only replicate frequencies below 22kHz and even that's within the hearable range of many (mostly younger) people.
I just really don't understand this widespread need to believe "lossy" doesn't really mean "lossy." It's a psychoacoustic model and people are different, and further you can (as demonstrated by the article) even show where various encoding artifacts are feasibly audible to plenty of people. I just don't get it.
I used to believe these things were audible until I got into A / B testing with people claiming to hear the difference. The vast majority cannot. I barely ever can. And that's when I'm focusing to try to detect a difference. It's not something I'll notice when listening for enjoyment. YMMV, but I highly suggest doing listening tests yourself.
Ha. That was interesting. I got 1/6, though out of the 5 I got wrong, I picked the 320Kbps MP3 for 4 of them. This backs up a few blind tests I did when I was younger, though then with music I was familiar with: 320Kbps MP3 and uncompressed WAV were typically indistinguishable to me, and, when they were audibly different, I still couldn't say which was better and which was worse.
The one I got completely wrong was the Jay Z track. Even re-listening knowing which is which, I still can't tell the difference :( - but this is a style of music I rarely listen to, and maybe that's the reason.
It's also a fair few years since I was 30, and that could be an alternative explanation.
(I'm actually quite surprised I did as well as I did, as the 128Kbps versions weren't obviously worse, and it did take a few listens on headphones. I should probably do another blind test with a 2003 build of LAME or whatever, not so much to compare MP3 vs WAV, but more to measure how much my hearing has deteriorated over the years...)
This is apparently a thing, that people will prefer what they’re used to and these days that’s low bitrate audio played through phone and laptop speakers...
These aren't songs I normally listen to, but I did this quiz just now and even with laptop speakers I picked 3x uncompressed, 2x 320kbps MP3, and 1x 128kbps MP3.
128 is usually pretty easy to pick out, once you know what to listen for. There's no soundstage. They just sound flat.
The tricky part is distinguishing between 320 and uncompressed. 320 is pretty darn good. Even when I can hear the difference, if it's not a song I'm familiar with, I can't tell which one is better. It's just different.
I've conducted several A/B tests back when I had nothing better to do and it was a very education experience. I used a recording of a harpsichord, using lame V5 mp3s, and came away with a statistically significant result (I remember taking 14 trials, though I don't remember what my actual true positive rate was)
After all of that, though I had double blind evidence I could hear a difference, I knew I didn't really care about that difference. I couldn't even describe the difference - it was almost subliminal. And that was with a harpsichord, which is a known difficult sound to encode. I've been a lot more comfortable knowing lossy encoding isn't destroying my music enjoyment since those tests.
There are different things that could be found through tests, but I don't think I've seen tests being really extensive.
The interesting questions to me:
- Can a difference be heard?
- If yes, are people able to reliably identify A vs B (as in, can people identify whether it's A or B rather that whatever they listen now is not the same as the last they heard)
- If they can identify a difference, which sounds better to them?
The answer to the last two questions is easy - yes to both.
There are absolutely huge differences between budget consumer and high-end professional equipment. There are also huge differences between high-end audiophile and professional equipment - the former being designed to flatter and enhance the sound, the latter being as neutral as possible.
A good few years ago a high-end audiophile CD/DAC maker was caught deliberately rolling off the top end on their equipment to make it sound smoother and "less digital."
It wasn't accurate, but a lot of people liked the sound enough to pay five figures for the boxes they made.
Professional audio has a different goal. The aim is to create a mix that "translates" - sounds good on as many different systems as possible, from budget earbuds to audiophile.
For a long time studios used a pair of shitty Yamaha bookshelf speakers called NS10s next to their ultra-expensive reference speakers. NS10s were almost the worst speaker ever. But if a mix sounded good on the NS10s, it sounded good on anything.
Blind testing is good for the first two questions, but it's next to impossible to blind test headphones.
The question of preference, though, is what matters, and that's where I am now firmly convinced lossy compression is fine for me: the sort of differences that I could hear made zero difference to my enjoyment of actual music.
I did these kinds of tests in my early 20s (not sure how it would go now) and completely destroyed them. (I shut many many people up doing these tests, actually.)
> I barely ever can.
> It's not something I'll notice when listening for enjoyment.
Some of the numbers in the article are unsourced and highly questionable. 130-140dB dynamic range of hearing? That's the difference between an anechoic chamber (~10-20dB) and enough sound pressure to literally rupture your eardrums (~150dB). Maybe you can technically perceive that dynamic range... once.
I certainly agree that artifacts in 128kbps MP3s are noticeable but the assertion that 320kbps AACs or MP3s created with modern encoders are audibly different from uncompressed audio really needs some evidence. Every single double blind test I have seen conducted between them has shown that nobody can tell.
The problem is people’s hearing range shrinks so much at low dB as to include it in ‘hearing range’ is meaningless. Similarly, people can detect pain at 140db, but they don’t differentiate sounds.
Ex at 60db detecting 100hz sounds is easy, at 20db it’s outside of human hearing range.
The question isn't whether lossy means "lossy," the question is whether lossy means "audibly different." Years ago Stereophile published an article called "MP3 vs AAC vs FLAC vs CD" which was really meant to argue for the superiority of lossless compression, but actually does a pretty good job of explaining both subtle differences between AAC and MP3 at the same bit rate and how high bit-rate lossy encoding pushes artifacts down to inaudible levels. You can put up an AAC/MP3 frequency graph compared to a FLAC/CD frequency graph and have it look absolutely horrifying, until you notice that the junk on the AAC/MP3 chart is all at -100db and below. Unless your listening room is an anechoic chamber, you're just not going to hear it. (In most cases it's probably below the noise floor of your reproduction equipment.)
I often think I can hear the difference between lossless and lossy, but IIRC, the last time I did ABX testing on that, there was only 1 out of 5 tracks I tried where I could tell the difference at a statistically significant rate. (And I'm not sure I'd be able to repeat that.) The best argument I have for storing music in FLAC (or ALAC or some other open lossless format) is analogous to the argument for storing images in a lossless rather than lossy format if you have the space: if you ever have to transcode the file to something else, start with the highest quality possible.
> The author explains that it's a psychoacoustic model and then demonstrates how various artifacts in various waveforms are the result of compromises within this model.
The explanations aren’t themselves incorrect — there’s certainly low-pass filtering, pre- and post-echo artifacting and so on, and these reflect in the waveforms — but to reference a waveform and to say “this sounds bad because it looks like this” doesn’t really hold water. Visual differences are not necessarily audible differences, which is the fact psychoacoustic encoding leverages.
If an encoder’s design goal was to produce visually-comparable waveforms, a different approach would be used.
> In fact, CDs can only replicate frequencies below 22kHz and even that's within the hearable range of many (mostly younger) people.
There is no useful or meaningful content above ~18 kHz (aside from dither) on most recordings. There may be exceptions, but by and large, on real music recordings, the information above that frequency isn’t valuable, and is the among the easiest things to discard.
Again, I have no idea what you think we disagree about. Your reply, like most of the replies, is premised on the idea that yes you can hear the difference in some cases.
To say nothing of what audio compression does to music (Ref: loudness wars [0])
I was intrigued by this paragraph in the article:
>> Psychoacoustics is the study of how humans perceive sound, and it's relevant here because advocates of lossy data compression argue that when listening to CD-quality audio, it is impossible for our brains to perceive all the data reaching our ears. It is, therefore, unnecessary — the argument goes — to store and reproduce all of that data. But which data can be removed is another question, and this is why various psychoacoustic principles are exploited in different amounts by different perceptual audio coding algorithms.
I wonder if different people will perceive sound differently (in terms of frequency and dynamics) purely based on physiological/psychological traits?
I've wondered if some of the loudness wars are driven by an attempt to compensate for lossy compression? The wars haven't abated - the latest by Panic At The Disco is the loudest I've ever ripped.
I don't have that one so I can't compare. I certainly won't claim to have seen the worst, I can only relate a general observation that things aren't getting better.
I don't really understand how the lossy vs lossless argument turned into a chest-beating contest about whose ears are the best.
The point of lossless audio compression is archiving - knowing that you can make a first-generation encoding of the audio you have stored for any new device or format in future.
If you want to listen to your FLAC files, great, you're not losing anything by doing that - just don't try and tell me that it's an aesthetic choice that makes the slightest bit of difference to the experience.
> I don't really understand how the lossy vs lossless argument turned into a chest-beating contest about whose ears are the best.
It's just basic human nature. As soon as you say "you can't see/hear/taste/perceive the difference between X and Y" some dick-waver will come in and start saying they can do it. And of course, the first to admit they can't hear a difference loses, and then you end up with things like Monster Cables, and the Emperor's new hi-fi setup.
I've seen people seriously claim they can hear the difference between the same audio signal burned onto different brands of CD-Rs.
Many people don't realize that Bluetooth requires compression applied to audio too. Theoretically the latest revisions allow already compressed music in certain formats to be transmitted as-is without recompression, but I'm guessing the data paths to allow that to happen don't usually exist. Double compression will be far more damaging.
But does the entire chain end-to-end support sending a file unmodified? If for example there's a mixer involved so that your ringtone can interrupt your music, that music can't be piped through without decoding. Is the volume control handled in the headset or does it come earlier? That can't be applied without decoding either.
When no mixing is occurring I believe there's no decode/re-encode happening so that battery savings are realized. If there are multiple audio streams then the mixer kicks in and for that time period there would be re-encoding occurring but from an acoustic quality perspective that wouldn't matter since the multiple audio streams would already alter the perceived quality relative to the reference.
Doesn’t the equipment used for reproducing the soundstage matter in A/B testing? I haven’t done a rigorous test myself, because I’m comfortable having a predilection for high fidelity audio even if it isn’t demonstrably/empirically superior. But in my experience listening to music on a “audiophile” or “studio reference” setup, when the equipment is sufficiently sensitive and well engineered as to most accurately render the audio signal, I recall rather obviously noticing a difference in sound quality.
I can’t imagine if you compare a 192kbps MP3 with a DSD256 audio file played on reference monitors from a high performance DAC there’d be no audible difference. Personally, using a McIntosh integrated amp and Focal Utopia headphones the first time I listened to a DSD track, 1-bit word depth sampled 2.8M times per second, it was unlike anything I’d ever heard before. I acknowledge that’s not a direct comparison, but all i’m saying is maybe the lossy vs lossless test performed on a low fidelity audio chain where the signal is always meaningfully subject to harmonic distortion before you hear it is the reason most people can’t tell a difference?
Personally I have zero problems A/B-ing DSD, PCM and lossy formats on my reference system (headphones). I also know (award winning) people in the audio industry who can walk into a listening room and tell you where the dominate resonances are or tell you if a reconstruction filter is linear or minimum phase.
The idea you can take a group of non-professionals and run tests with no requirements on equipment quality and then draw conclusions about what any human can hear is absurd.
Those tests (Hydrogenaudio was one of the main proponents back when lossy codecs were more important) are a decent way to tell if something is audible for the average listener with average equipment. It cracks me up when people start telling me what I can or cannot hear based on those tests without knowing anything about me or what kind of equipment I own. This happened just the other day on an audio forum.
Another dimension to this is that DAC quality has been increasing steadily while the prices of high fidelity DACs have been dropping. The DAC chips on the market today are really the best ever made (ESS and AKM notably). More people than ever have access to (near-)reference quality DACs. When lossy compression tests were popular few people had access to reference quality DACs. I remember people talking about using their computer sound cards as sources for those tests.
I think a lot of people react negatively to this topic because it's considered elitist (expensive toys). The good news is just about anyone can afford a near-reference quality, inexpensive headphone setup these days. You have to do your homework and read reviews but they definitely exist. I recently picked up a DAP and IEMs for ~$300 total (for running) and it's 90% of the quality of my reference rig. I listen to it instead of my reference rig sometimes. It's that good, despite the price.
(Even on that, I can easily ABX FLAC and 192kps AAC, which has been universally declared "transparent" more than once.)
If you think your smartphone's audio jack sounds good, you really need to listen to a device with a proper amount of output power (you need more than you think for headphones) and a high resolution DAC. You need both to be good, though they can be part of the same device. When you have them, music really becomes holographic and, for lack of a better word, alive.
I've got a lot of different gear, but the stuff in question was the HiBy R3, which has a new ESS DAC/AMP combo (an SoC which eliminates a lot of variability in quality due to implementation problems) and a very healthy 200mW/16R (or somewhere around there) output power in balanced mode. Small, the software is great by DAP standards and it's inexpensive.
The IEMs I struggle to recommend: Fiio FH1, which is a dynamic driver / BA 'hybrid'. They are enjoyable to listen to but definitely not reference -- bass heavy. They're leagues beyond a typical earbud, however, and for $75 it's hard to complain.
I mainly don't recommend them because they're the first IEMs I've tried anywhere near that price range, apart from freebies included with smartphones, etc. They might be easily bettered by something else in the same price range, especially since there's a lot of competition there now.
If anyone is looking for a low cost, open source, high quality Headphone amp + DAC combo which rivals gear costing 16 times as much, take a look at NwAvGuys stuff.
>It cracks me up when people start telling me what I can or cannot hear based on those tests without knowing anything about me or what kind of equipment I own. This happened just the other day on an audio forum.
Just disprove them with data! Until then, casually mentioning "award-winning" acquaintances and telling people they need "proper" gear reads a bit condescending and probably turns some people defensive.
I generally don't name drop and I never tell people to get better gear. The burden of proof is not on me. I simply do not give a shit what they believe.
I only take issue with people telling me what I do or do not experience -- because of the absurdity of that. The fact that this is about audio is coincidental.
If they tell you this in a personal dialogue, than I concur, it's silly and probably pointless. However, in a public online discussion they are probably just trying to mark for others some information which they see as false as something to disregard. And if they're reasonable, they will add some statistical data to prove their point.
The impact of audio hardware quality specifically on the noticeability/ABXability of compression artifacts is greatly exagerated. Going above mid-range speaker setups I have not found them to be more revealing in this specific regard. And of course, as you mentioned, even inexpensive but decent headphones beat very expensive speaker setups for critical listening.
I completely missed the Hydrogenaudio multiformat test in 2014. I wonder how much things progressed since when I participated in the early tests in 2006.
There's one point i've read somewhere that really made methink differently on this subject: It was claiming that listening to lossy formats will increase brain activity, because you're brain makes up for some missing information on thy fly. This is not a conscious proccess. But after listening to music for a while, lossless music could be less exhausting and more relaxing because of that.
Not completely sure, if this is true, I couldn't find the study for it right away.
I would believe that some of the distortion on the edge of perception might cause a little bit of stress (like, say, the barely audible whine of bad fluorescent lighting), but not really the bit about the brain working harder.
Compared to the many anti-scientific audiophile articles, this one was quite decent.
A bit that bothered me was the description of how the encoders work. It mentions the discarding of frequencies, which is not entirely correct. The only real discarding of frequencies happens with the lowpass filter that is typically applied as a pre-processing step, not the actual encoding.
The encoding of the frequency components happens with coefficients, which are quantized (coarsely encoded), not zeroed. This introduces noise to the signal, which is weighted against the computed masking threshold, driving bit allocation to the different frequency bands.
As for the format recommendation for people ripping their own CDs, I'd say stick to a lossless format for storage at home. That way you're future-proof. For mobile use, make encodings from those files using whatever best lossy format is available at the time.
This is what I do at home. I rip CDs using EAC (Exact Audio Copy) to FLAC with a .CUE file. That way I can recreate the image if needed. I usually upload the FLAC to Google Music to add to my library for my portable needs.
Compression glitches can also produce beautiful harmonics as samples' quality degrades: Mille Plateaux just released Bienoise's "Most beautiful design"[0], an LP that can fit on a 1.44 floppy disk thanks to mp3 compression and clever mixing
58 comments
[ 3.6 ms ] story [ 132 ms ] threadThis is a somewhat older article, but most lossy encoders at the time were still very good even at fairly low (sub-96 kbps) bit rates.
I just really don't understand this widespread need to believe "lossy" doesn't really mean "lossy." It's a psychoacoustic model and people are different, and further you can (as demonstrated by the article) even show where various encoding artifacts are feasibly audible to plenty of people. I just don't get it.
The one I got completely wrong was the Jay Z track. Even re-listening knowing which is which, I still can't tell the difference :( - but this is a style of music I rarely listen to, and maybe that's the reason.
It's also a fair few years since I was 30, and that could be an alternative explanation.
(I'm actually quite surprised I did as well as I did, as the 128Kbps versions weren't obviously worse, and it did take a few listens on headphones. I should probably do another blind test with a 2003 build of LAME or whatever, not so much to compare MP3 vs WAV, but more to measure how much my hearing has deteriorated over the years...)
128 is usually pretty easy to pick out, once you know what to listen for. There's no soundstage. They just sound flat.
The tricky part is distinguishing between 320 and uncompressed. 320 is pretty darn good. Even when I can hear the difference, if it's not a song I'm familiar with, I can't tell which one is better. It's just different.
After all of that, though I had double blind evidence I could hear a difference, I knew I didn't really care about that difference. I couldn't even describe the difference - it was almost subliminal. And that was with a harpsichord, which is a known difficult sound to encode. I've been a lot more comfortable knowing lossy encoding isn't destroying my music enjoyment since those tests.
The interesting questions to me:
- Can a difference be heard?
- If yes, are people able to reliably identify A vs B (as in, can people identify whether it's A or B rather that whatever they listen now is not the same as the last they heard)
- If they can identify a difference, which sounds better to them?
- Do different headphones give different results?
- Same question with headphones vs speakers?
There are absolutely huge differences between budget consumer and high-end professional equipment. There are also huge differences between high-end audiophile and professional equipment - the former being designed to flatter and enhance the sound, the latter being as neutral as possible.
A good few years ago a high-end audiophile CD/DAC maker was caught deliberately rolling off the top end on their equipment to make it sound smoother and "less digital."
It wasn't accurate, but a lot of people liked the sound enough to pay five figures for the boxes they made.
Professional audio has a different goal. The aim is to create a mix that "translates" - sounds good on as many different systems as possible, from budget earbuds to audiophile.
For a long time studios used a pair of shitty Yamaha bookshelf speakers called NS10s next to their ultra-expensive reference speakers. NS10s were almost the worst speaker ever. But if a mix sounded good on the NS10s, it sounded good on anything.
The question of preference, though, is what matters, and that's where I am now firmly convinced lossy compression is fine for me: the sort of differences that I could hear made zero difference to my enjoyment of actual music.
> I barely ever can. > It's not something I'll notice when listening for enjoyment.
Pretty much tell the story.
I certainly agree that artifacts in 128kbps MP3s are noticeable but the assertion that 320kbps AACs or MP3s created with modern encoders are audibly different from uncompressed audio really needs some evidence. Every single double blind test I have seen conducted between them has shown that nobody can tell.
I can most certainly tell the difference between 320kbps lossy and red book audio on well done recordings.
Ex at 60db detecting 100hz sounds is easy, at 20db it’s outside of human hearing range.
I often think I can hear the difference between lossless and lossy, but IIRC, the last time I did ABX testing on that, there was only 1 out of 5 tracks I tried where I could tell the difference at a statistically significant rate. (And I'm not sure I'd be able to repeat that.) The best argument I have for storing music in FLAC (or ALAC or some other open lossless format) is analogous to the argument for storing images in a lossless rather than lossy format if you have the space: if you ever have to transcode the file to something else, start with the highest quality possible.
Hate to sound like a broken record, but what are you arguing with me about?
The explanations aren’t themselves incorrect — there’s certainly low-pass filtering, pre- and post-echo artifacting and so on, and these reflect in the waveforms — but to reference a waveform and to say “this sounds bad because it looks like this” doesn’t really hold water. Visual differences are not necessarily audible differences, which is the fact psychoacoustic encoding leverages.
If an encoder’s design goal was to produce visually-comparable waveforms, a different approach would be used.
> In fact, CDs can only replicate frequencies below 22kHz and even that's within the hearable range of many (mostly younger) people.
There is no useful or meaningful content above ~18 kHz (aside from dither) on most recordings. There may be exceptions, but by and large, on real music recordings, the information above that frequency isn’t valuable, and is the among the easiest things to discard.
I was intrigued by this paragraph in the article:
>> Psychoacoustics is the study of how humans perceive sound, and it's relevant here because advocates of lossy data compression argue that when listening to CD-quality audio, it is impossible for our brains to perceive all the data reaching our ears. It is, therefore, unnecessary — the argument goes — to store and reproduce all of that data. But which data can be removed is another question, and this is why various psychoacoustic principles are exploited in different amounts by different perceptual audio coding algorithms.
I wonder if different people will perceive sound differently (in terms of frequency and dynamics) purely based on physiological/psychological traits?
[0] - https://en.wikipedia.org/wiki/Loudness_war
The point of lossless audio compression is archiving - knowing that you can make a first-generation encoding of the audio you have stored for any new device or format in future.
If you want to listen to your FLAC files, great, you're not losing anything by doing that - just don't try and tell me that it's an aesthetic choice that makes the slightest bit of difference to the experience.
It's just basic human nature. As soon as you say "you can't see/hear/taste/perceive the difference between X and Y" some dick-waver will come in and start saying they can do it. And of course, the first to admit they can't hear a difference loses, and then you end up with things like Monster Cables, and the Emperor's new hi-fi setup.
I've seen people seriously claim they can hear the difference between the same audio signal burned onto different brands of CD-Rs.
I can’t imagine if you compare a 192kbps MP3 with a DSD256 audio file played on reference monitors from a high performance DAC there’d be no audible difference. Personally, using a McIntosh integrated amp and Focal Utopia headphones the first time I listened to a DSD track, 1-bit word depth sampled 2.8M times per second, it was unlike anything I’d ever heard before. I acknowledge that’s not a direct comparison, but all i’m saying is maybe the lossy vs lossless test performed on a low fidelity audio chain where the signal is always meaningfully subject to harmonic distortion before you hear it is the reason most people can’t tell a difference?
The idea you can take a group of non-professionals and run tests with no requirements on equipment quality and then draw conclusions about what any human can hear is absurd.
Those tests (Hydrogenaudio was one of the main proponents back when lossy codecs were more important) are a decent way to tell if something is audible for the average listener with average equipment. It cracks me up when people start telling me what I can or cannot hear based on those tests without knowing anything about me or what kind of equipment I own. This happened just the other day on an audio forum.
Another dimension to this is that DAC quality has been increasing steadily while the prices of high fidelity DACs have been dropping. The DAC chips on the market today are really the best ever made (ESS and AKM notably). More people than ever have access to (near-)reference quality DACs. When lossy compression tests were popular few people had access to reference quality DACs. I remember people talking about using their computer sound cards as sources for those tests.
I think a lot of people react negatively to this topic because it's considered elitist (expensive toys). The good news is just about anyone can afford a near-reference quality, inexpensive headphone setup these days. You have to do your homework and read reviews but they definitely exist. I recently picked up a DAP and IEMs for ~$300 total (for running) and it's 90% of the quality of my reference rig. I listen to it instead of my reference rig sometimes. It's that good, despite the price.
(Even on that, I can easily ABX FLAC and 192kps AAC, which has been universally declared "transparent" more than once.)
If you think your smartphone's audio jack sounds good, you really need to listen to a device with a proper amount of output power (you need more than you think for headphones) and a high resolution DAC. You need both to be good, though they can be part of the same device. When you have them, music really becomes holographic and, for lack of a better word, alive.
Which ones did you choose?
The IEMs I struggle to recommend: Fiio FH1, which is a dynamic driver / BA 'hybrid'. They are enjoyable to listen to but definitely not reference -- bass heavy. They're leagues beyond a typical earbud, however, and for $75 it's hard to complain.
I mainly don't recommend them because they're the first IEMs I've tried anywhere near that price range, apart from freebies included with smartphones, etc. They might be easily bettered by something else in the same price range, especially since there's a lot of competition there now.
Headphone amp: http://nwavguy.blogspot.com/2011/08/o2-details.html#resource...
DAC: http://nwavguy.blogspot.com/2012/04/odac-released.html
Just disprove them with data! Until then, casually mentioning "award-winning" acquaintances and telling people they need "proper" gear reads a bit condescending and probably turns some people defensive.
I only take issue with people telling me what I do or do not experience -- because of the absurdity of that. The fact that this is about audio is coincidental.
I completely missed the Hydrogenaudio multiformat test in 2014. I wonder how much things progressed since when I participated in the early tests in 2006.
Not completely sure, if this is true, I couldn't find the study for it right away.
And it makes data compression worse, because when EVERYTHING is loud, sounds that should have been masked by others are not.
https://benchmarkmedia.com/blogs/application_notes/intersamp...
A bit that bothered me was the description of how the encoders work. It mentions the discarding of frequencies, which is not entirely correct. The only real discarding of frequencies happens with the lowpass filter that is typically applied as a pre-processing step, not the actual encoding. The encoding of the frequency components happens with coefficients, which are quantized (coarsely encoded), not zeroed. This introduces noise to the signal, which is weighted against the computed masking threshold, driving bit allocation to the different frequency bands.
As for the format recommendation for people ripping their own CDs, I'd say stick to a lossless format for storage at home. That way you're future-proof. For mobile use, make encodings from those files using whatever best lossy format is available at the time.
[0](https://forceincmilleplateaux.bandcamp.com/album/most-beauti...)