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Listen to what gets lost when an MP3 is made (2015)

142 points by teleforce 2y ago

Timwi 2y ago
I would have liked a comparison with Ogg and perhaps other formats. I hear a lot about MP3 throwing away a lot more than Ogg but I'd love to see real data on it.
- arch1t3cht 2y ago
  ogg is a container format, not a coding format.
  - wizzwizz4 2y ago
    Vorbis, then.
- stavros 2y ago
  I don't think the claim is that MP3 throws a lot more away, it's that Vorbis sounds better. The two are very different, Vorbis might be throwing a lot more away, but if it's only the stuff you can't hear, then it'll sound better.
Klaster_1 2y ago
The article doesn't mention at what bit-rate the difference track was made, anyone knows? Seems disingenuous and pro-"authentic" otherwise.
- rob74 2y ago
  If it's really the original MP3 version of Tom's Diner produced by the Fraunhofer engineers, probably not a very high bitrate. Aside from that, I would say that even back in 2015, MP3 was already on its way out and replaced by better (while still lossy) compression methods?
  - porbelm 2y ago
    With space not much of an issue anymore, FLAC is pretty much the default nowadays. And even though Opus and AAC and others have better encoding than old MP3, but I guess a 128 Kbps MP3 encoded in the original Fraunhofer l3enc (the best back then) and one encoded with LAME will be different - and the LAME version will be "better" because of improvements in psychoacoustics? At least I remember l3enc being MUCH better than anything else at 128 Kbps (Xing lol, cymbal washing anyone?) before LAME came along.
    - klez 2y ago
      > With space not much of an issue anymore, FLAC is pretty much the default nowadays The default for what? Space is not the only consideration. What about bandwidth? I'm pretty sure spotify, deezer and the others are not transmitting FLACs, especially not at the base quality level.
- Sesse__ 2y ago
  It also doesn't really mention how this “lost” material was identified. If you just subtract the encoded from the original, then any phase difference will make it sound like material “disappeared”, while in reality, it just came very slightly earlier or later.
  - rob74 2y ago
    I guess it was exactly as you write - but instead of slightly earlier or later, the "lost" sounds are the high frequencies (a lot of hissing, clicking etc.) - the actual sound is mostly still there, but slightly "muffled" because it contains only the lower-frequency components.
- etwas 2y ago
  It's 128kb. The useful and informative article is mentioned at the end: http://theghostinthemp3.com/theghostinthemp3.html http://theghostinthemp3.com/theghostinthemp3.html
0points 2y ago
This is why we dont encode mp3 in 96kbps or whatever.
CGamesPlay 2y ago
Found the original author's page about the project (no longer on the internet): https://web.archive.org/web/20211011015410/http://ryanmaguiremusic.com/theghostinthemp3.html https://web.archive.org/web/20211011015410/http://ryanmaguir... One interesting thing to note: this is a composition, not an analysis. It's not fully documented exactly what modifications to the "raw data" were made.
- roelschroeven 2y ago
  I can open the original link, https://www.theghostinthemp3.com/theghostinthemp3.html https://www.theghostinthemp3.com/theghostinthemp3.html, just fine. Maybe it's reinstated? Or there was a temporary problem?
  - deleted 2y ago
    [deleted]
jonathanstrange 2y ago
So what. People listened to music on mechanical gramophones and enjoyed it. Too many audio engineers think it's all about the sound, when in the end it's about the music and the feelings it expresses.
- a-french-anon 2y ago
  You're naïve if you think that 1) Good enough prevents better from existing for some people, 2) Technological advances had the same consequences for all material. For example, orchestral music massively benefited from CD's increased dynamic range.
- lukan 2y ago
  "People listened to music on mechanical gramophones and enjoyed" Herman Hesse (or rather Harry Haller in Steppenwolf) used to get enraged by that - distorted garbage and a blasphemy to the godly componists. But then eventually he overcame it because of exactly that reasoning - if people enjoy it and are touched by the music - it works good enough. That is the purpose of music, not arbitrary perfectionism.
- sevensor 2y ago
  The way I think about recorded music is, the instrument being played is ultimately a set of speakers. That’s a musical experience to be encountered on its own terms. Live music is a wonderful thing, and presenting an illusion of live performance is certainly one thing a recording / reproduction system can do, but that doesn’t mean fidelity is the only thing we value when listening to a recording. Consider the punk music fans who enjoy a cheap stretched out cassette tape; I’m not going to deny them their enjoyment any more than I’ll deny the golden eared audiophile. They enjoy listening to different instruments. And this brings me to my point: the engineer is no mere technician whose job is only fidelity. The engineer is part of the performance through speakers, an artist no less than the musicians being recorded.
  - mrob 2y ago
    Even the best stereo speakers have an inherent weakness: comb filtering between the channels. You can easily test this yourself by generating some mono pink noise in Audacity (or your favorite audio editor) and playing it on stereo speakers. Move your head and you will hear changes in the sound like a flanger effect. This is most obvious if you are close to the speakers and in an acoustically dry environment. Compare with the same sound hard panned to a single speaker. This is one reason why a real physical center channel improves clarity for movie dialogue.
a-french-anon 2y ago
This article could at least have a paragraph explaining (in a dumbed-down way) why various kinds of psychoacoustic masking (temporal, frequency) make what's removed almost inaudible anyway. Reading the linked source (https://www.theghostinthemp3.com/theghostinthemp3.html https://www.theghostinthemp3.com/theghostinthemp3.html), he at least used LAME, but at a fixed 128 kbps bitrate, not in VBR mode =( EDIT: nerds should read about sfb21 (https://wiki.hydrogenaud.io/index.php?title=LAME_Y_switch https://wiki.hydrogenaud.io/index.php?title=LAME_Y_switch), AAC, Vorbis and Opus (CELT) aren't just theoretical improvements
- magicalhippo 2y ago
  IIRC then 128kbps and lower means a 16kHz low-pass filter is applied, while it's not for higher bitrates. So in that case not just psychoacoustics.
  - theandrewbailey 2y ago
    I was under the impression that MP3 can't represent any frequency above 16kHz, no matter the bitrate.
    - gwbas1c 2y ago
      Then they would all be 32khz, instead of 42.1khz or 48khz.
    - Sonic656 2y ago
      MP3 is only good at 18KHz at V0/320kbps, Yet AAC & Vorbis can handle full 22KHz at 256kbps VBR with hotter audio(-15 ~ -5db) their VBR models can just use 384 ~ 960kbps. Yes Vorbis can actually reach 1.35mbit since It has no frame cap.
Quarondeau 2y ago
Interesting approach. So are we only able to hear those sounds now because the rest of the music was removed, which would ordinarily mask the missing sounds? To say that the mp3-encoded version is not "what the artist recorded and wanted for us to hear" would imply that we can hear all sounds in the uncompressed recording.
- varjag 2y ago
  Yes exactly this. On a high bitrate stream these losses are imperceptible (yes even by you golden ears out here) due to how auditory filters work in human hearing.
  - loa_in_ 2y ago
    That's assuming we understand human auditory filters AND human beings have uniform auditory filters all around
    - skrebbel 2y ago
      You can test this. Get a bunch of audiophiles in a room, play various recordings from various media (24KHz wav, mp3 at good bitrates and encoding settings, bad mp3, ogg, CD, vinyl etc) without showing them which is which, ask which one they like the most and see if the results correlate with the supposed quality of the source. I don't have a source ready but this has been a hot topic in audiophile land for decades and tldr is they'll pick out the really bad sources (eg <128kbps mp3) but not the rest. Basically the results look like those from a blind beer tasting test: no correlation between winner and supposed quality, except if the quality is especially bad. I'm no scientist, but to me "audiophiles who really care about this stuff can't pick out the good MP3 from the uncompressed original" is sufficient proof that MP3 is, actually, based on a sufficiently well-understood model of human hearing.
    - mrob 2y ago
      We don't need to understand those things to determine if the differences are audible. We can simply perform listening tests: https://en.wikipedia.org/wiki/ABX_test https://en.wikipedia.org/wiki/ABX_test A properly conducted ABX test is the most favorable condition possible for detecting a difference. If you can't ABX it you can't hear it. There's an ABX testing website with various lossy formats you could try: https://abx.digitalfeed.net/list.html https://abx.digitalfeed.net/list.html However, failing to ABX those specific samples does not guarantee you are unable to tell the difference in all circumstances. There are some sounds that are unusually difficult to encode ("killer samples"). This is an especially big problem for MP3. The LAME project has a collection of killer samples for MP3: https://lame.sourceforge.io/quality.php https://lame.sourceforge.io/quality.php More modern lossy formats are less susceptible to killer samples, but theoretically there could still be problematic cases.
NoPicklez 2y ago
Fairly lackluster article. Not all .mp3's are created equally and can vary in how lossy they are based on the bitrate. If you care enough to want to hear exactly what the artist wants you to hear, you just listen to the lossless version.
- flanked-evergl 2y ago
  The "lossless" version is in fact not entirely lossless. There is much that gets lost in the process of digitization, and even if there is no digitization happening the analogue equipment has some frequency response range.
  - vinhcognito 2y ago
    Maybe theoretically but for our purposes doesn't the Nyquist–Shannon sampling theorem mean it's essentially lossless to our ears?
- sumtechguy 2y ago
  > Not all .mp3's are created equally No kidding. There are a decent number of options to pick for quality of mp3. VBR/CBR, bitrate, joint/stereo/mono, and so on. I personally just pick something that sounds fairly close to the original. But that only really matters for me when it is side by side. Give me a few days between picking and I can not really tell anymore.
Traubenfuchs 2y ago
...I think this person just created a new genre of music. Something like: "What's lost noise." I immensely enjoyed listening to the "lost material" of Tom's Diner and would like to hear more of this! Maybe one could diff with a lower quality version, one where more has been cut away, more is lost/left over? There are so many possibilities!
- JKCalhoun 2y ago
  Lost-Fi.
  - Traubenfuchs 2y ago
    That's the Zeitgeist.
reliablereason 2y ago
You can try it yourself: ffmpeg -i original.wav -codec:a libmp3lame -b:a 192k output.mp3 && \ ffmpeg -i output.mp3 decoded.wav && \ ffmpeg -i original.wav -i decoded.wav -filter_complex "[1:a]aresample=async=1,volume=-1.0[inverted];[0:a][inverted]amix=inputs=2:weights=1 1" difference.wav
- schroffl 2y ago
  This produces an error in version 7.1 for me: [AVFilterGraph @ 0x6000008b59d0] More input link labels specified for filter 'aeval' than it has inputs: 2 > 1 [AVFilterGraph @ 0x6000008b59d0] Error linking filters Failed to set value '[0:a][1:a]aeval=val(0)-val(1):c=same' for option 'filter_complex': Invalid argument Error parsing global options: Invalid argument
  - atoav 2y ago
    The theory behind it is simple: Subtract each audio sample in B from each audiosample in A. You can do the same thing in your DAW¹ by putting A (e.g. the original) onto one channel and B (the processed sound) onto another. Then you invert the phase of B and listen to/export the sum. This trick works also for audio gear that claims it does amazing things to your sound (here you just need to make sure to match the levels if they have been changed). Then you can look how much of the signal has truly been affected by your 1000 bucks silver speaker cable. ¹ Digital Audio Workstation, something as simple as Audacity should do the trick
    - TheRealDunkirk 2y ago
      > Then you can look how much of the signal has truly been affected by your 1000 bucks silver speaker cable. I have a friend who has spent ridiculous sums of money on audio gear. Like, he's in his 50's, and still lives with his parents (in part) because of it. Over the years, I've learned I will never convince him that he's being fleeced, but I've wanted to make a site to host such A/B comparisons for a very long time, to perhaps get through to others what a waste most of the "audiophile" gear is.
ezconnect 2y ago
When I first experience CD audio it was too high pitch compared to tape versions. MP3 came along and each song sound different depending on the MP3 compression settings.
- jampekka 2y ago
  Some CDs were mastered to have pre-emphasis that boosted high frequencies if the player didn't properly account for it. This caused them to sound "high pitched". https://wiki.hydrogenaud.io/index.php?title=Pre-emphasis https://wiki.hydrogenaud.io/index.php?title=Pre-emphasis
BurpyDave 2y ago
Ironically, the 'diff' is compressed anyway, because it's on Vimeo, so that's not the actual diff either!
- taneq 2y ago
  This is like those articles on 10-bit colour where they show you "how vibrant and rich" the higher bit depth is, but you're reading it on the same old 8-bit-per-channel monitor.
HPsquared 2y ago
This could also be done on visual compression with JPEGs. Or on video compression, for that matter. It just shows though that these diffs are invisible to a human - by design.
- amelius 2y ago
  Animals and aliens might cringe at these images and sounds, though. ps, you could do the same thing with watermarked content.
moomin 2y ago
I think the thing that's really sticks out is that the breath noise are gone, which is one of the things that gives the track its character. Willing to bet the same kind of thing happens to fret noise as well.
- Fade_Dance 2y ago
  My headphone rig was always optimized for a nice guitar sound, and I would second that. The sound of fingers on the string and the "pluckiness" of the guitar is what gets lost. Beyond that, the specific thing that i noticed gets lost is bass character on some tracks. Ex: Some drum and bass tracks just don't hit at low bitrate. This aspect sometimes feeds into the low guitar strings though, where they might have a bit less body. Lastly is of course sound staging, but that's something that a headphone setup is very sensitive to. As for quality differences, I basically fall in line with the consensus on this thread. FLAC and 320 are indistinguishable. 192khz is almost always indistinguishable and good enough, although there are some situations where it might be slightly noticeable. 128 is pretty easy to tell the difference with the good setup. There is also the rare track with amazing production and or very cool stuff happening somewhere in the spectrum, so I don't entirely write off someone wanting a plus one on the take above. "I absolutely love this jazz album. It's been a large part of my musical journey as a human, I can a/b test this at 320, but for this album, I really want it lossless." I can respect that. I got to know some of my test tracks pretty well (ex Black Sun Empire & Arrakis for bass), and while 192 was fine for other tracks, I wanted 320 or lossless on those.
  - Optimal_Persona 2y ago
    As a musician recording ideas on a handheld Zoom recorder so I don't forget them, I switched from WAV to 320 MP3 a few years ago to save on file space and don't regret it.
no-such-address 2y ago
Funny article. "The exact master copy that the artist recorded and wanted for us to hear" In the digital era, does that even, uniquely, exist? "a set of standards created by a bunch of engineers in 1993" Nice! Was hoping the article would mention double blind studies about the ability to perceive differences and the quality between various audio file format, available elsewhere. Interesting, though not as overwrought as the reporting in this article.
- chasil 2y ago
  In fact, Glenn Gould's final recording of Bach's Goldberg Variations was rereleased from the analog tape recording, as the sound quality was better than digital of the time (1981). https://en.wikipedia.org/wiki/Bach:_The_Goldberg_Variations_(Glenn_Gould_album)#Re-releases https://en.wikipedia.org/wiki/Bach:_The_Goldberg_Variations_...
cladopa 2y ago
There is a trick there. A sound can mask another sound. You will not be able to tell the difference with both sounds playing at the same time, but if you subtract them you can hear it because there is no masking. I always loved to test the ears of my "Audiophile" friends. They will tell you how different MP3s are. You make a bet they can not differentiate them in 20 trials better than chance. I won with most people but some professional musicians that can identify little differences.
- mrob 2y ago
  >A sound can mask another sound. Details: https://en.wikipedia.org/wiki/Auditory_masking https://en.wikipedia.org/wiki/Auditory_masking Bernhard Seeber of the Audio Information Processing Group at the Technical University of Munich has some good demonstration videos on Youtube: https://www.youtube.com/watch?v=R9UZnMsm9o8 https://www.youtube.com/watch?v=R9UZnMsm9o8 https://www.youtube.com/watch?v=bU0_Kaj7cPk https://www.youtube.com/watch?v=bU0_Kaj7cPk
- deleted 2y ago
  [deleted]
- maep 2y ago
  I spent half my professional career doing listening tests (MUSHRA and P.800), specifically on test items like Tom's diner. 128 kbps mp3 ist fairly easy to pick out, especially if you can compare it to the original. Double the bitrate and it's a real challenge. Modern codes like opus are much more efficient. At high bitrates they are fully transparent and anybody who claims to be able to hear a difference is full of shit. Put them in a controlled setting and they fail every time.
  - anotherhue 2y ago
    LAME @192kb/s VBR was transparent 20 years ago, that said FLAC is still a good choice because storage is cheap now and you don't want to have to deal with a copy-of-a-copy situation. Some Young folk think that 24bit/192kHz is the one-true-form who would think a 16/44 FLAC is a lossy encode, and then there's the vinyl folks. (I like vinyl, but not for the fidelity). Required reading: https://xiph.org/video/vid2.shtml https://xiph.org/video/vid2.shtml
grishka 2y ago
Now I want this comparison for Opus. It doesn't do that whole psychoacoustics thing, does it? But it also somehow manages to ~double the compression ratio compared to MP3 without any noticeable difference in the sound quality.
- bjoli 2y ago
  I haven't gotten into reading about audio and video compression yet, but it from where I am standing now it really looks like magic.
gwbas1c 2y ago
In 1999 when MP3 was getting attention, I tried to do this. I encoded a file, then inverted it, and mixed it back into the original. It didn't cancel anything out. The reason: Mp3 dramatically alters phase. Because all the phases are different, it's hard to naively determine how the signal is altered. Years later, I took the time to write a series of tools to investigate lossy audio: https://andrewrondeau.com/blog/2016/07/deconstructing-lossy-audio-the-case-for-lossless https://andrewrondeau.com/blog/2016/07/deconstructing-lossy-...
- tecleandor 2y ago
  Oh! Is that an artifact of Joint Stereo encoding ?
  - gwbas1c 2y ago
    > Oh! Is that an artifact of Joint Stereo encoding ? I personally don't understand enough of MP3's internals to explain that. What I assume is that, because MP3 internally stores Fourier transforms in the frequency domain, instead of the time domain, it uses very few bits to store phase. This will result in phase shifts. Hopefully, someone else can give a better explanation of this than I can. Basically, think of it this way: 1: Imagine a frame of 64 16-bit samples. (1024 bits) 2: Than its converted to 32-bit floats: 64 32-bit samples. 3: Than its transformed 4: Now there are: 1 DC bias 32-bit float, where phase is 0, and 32 frequency amplitude floats, and 32 phase floats. (The other 31 frequency/phase pairs are duplicates and don't need to be stored.) These floats need to be quantized so that there are many less bits. Some of it can happen by using very few bits to store phase, and some of it can happen by using less bits to store amplitudes. Again, someone else can probably explain this better than I can.
    - tecleandor 2y ago
      Interesting. In a quick internet search seems like people finds a small phase shift on MP3 enconding, but I haven't found details on why. Haven't had much time to find it, though...
sdk77 2y ago
Very interesting! The audio of Tom's Dinner rejected by the encoding sounds mesmerizing to me. I still find it to be musical - it reminds me of a record I bought a really long time ago, it was called modulation & transformation on mille plateaux, it's a collection of songs in the abstract and experimental genre.
jonnycomputer 2y ago
"What MaGuire has proved here is that the songs we listen to every single day are not the exact master copy that the artist recorded and wanted for us to hear. Instead, they are slightly stripped versions of their art run through a set of standards created by a bunch of engineers in 1993. For many people, that won’t matter. The songs sound almost the same, but the compression of music into an MP3 format is an important question to weigh when considering artistic intent and analyzing songs that aren’t exactly the original." I feel like this analysis isn't well grounded in what artists and sound engineers actually do, or how they think.
- chasil 2y ago
  When I've used lame --preset extreme, it uses variable bit rate which was developed some time later.
- Optimal_Persona 2y ago
  I honestly didn't know that was a matter of contention at this point...it's very common for pro (or smart amateur) audio engineers to actually listen to the effect of different digital compression algorithms and adapt accordingly. Things like Sonnox Fraunhofer plugin let you do this. https://sonnox.com/products/oxford-fraunhofer-pro-codec https://sonnox.com/products/oxford-fraunhofer-pro-codec - In this month's TapeOp magazine, Jeff Jones mentions he always monitors through such a codec so he can adjust accordingly for the best sound regardless of final format https://tapeop.com/interviews/163/jeff-jones/ https://tapeop.com/interviews/163/jeff-jones/ - IMO Rush's Moving Pictures is one of the best (and best-sounding) albums from the '80s despite the fact that it was mastered to a Sony digital device that's renowned as terrible sounding, with only 14 bits of usable resolution. How/Why? The production team monitored through the Sony system while mastering and made tweaks as they went to account for its limitations. - I've had my own music professionally mastered at high resolution, and discovered that converting hi-res to MP3 (even at highest bitrate) caused digital peaks over 0.0 dB, I fixed by normalizing to -0.3 dB. TBH if the resulting file is going to be transmitted over BlueTooth it's going to be further degraded. And yet people can still make a strong emotional connection to the underlying music and artist...
pvillano 2y ago
Two instances where lossy compression failed for me are the movie Koyaanisqatsi and songs by the artist TOBACCO. Koyaanisqatsi has a lot of film grain and TOBACCO uses a lot of distortion. There is noise in there, but it's very deeply mixed into the signal.
ipunchghosts 2y ago
There's still audio motifs in there that can be further optized out. If the remaining audio was noise like, I would say we reached the compression limit.
Agraillo 2y ago
Why Tom's Dinner? Because it is a cappella. There's a book "How Music Got Free" by Stephen Witt [1] detailing the history of mp3 format and related events. It is a very good read and there's an explanation Increases in processing power spurred progress. Within a year Brandenburg’s algorithm was handling a wide variety of recorded music... But one audio source was proving intractable: what Grill, with his imperfect command of English, called “the lonely voice.” (He meant “lone.”) Human speech could not, in isolation, be psychoacoustically masked. Nor could you use Huffman’s pattern recognition approach—the essence of speech was its dynamic nature, its plosives and sibilants and glottal stops. Brandenburg’s shrinking algorithm could handle symphonies, guitar solos, cannons, even “Oye Mi Canto,” but it still couldn’t handle a newscast. Stuck, Brandenburg isolated samples of “lonely” voices. The first was a recording of a difficult German dialect that had plagued audio engineers for years. The second was a snippet of Suzanne Vega singing the opening bars of “Tom’s Diner,” her 1987 radio hit. [1] https://en.wikipedia.org/wiki/How_Music_Got_Free https://en.wikipedia.org/wiki/How_Music_Got_Free
chrsgrrtt 2y ago
I developed a streaming service many years ago; Dolby wanted us to use their codec for the audio, and they used a track just like this as the primary basis of their sales pitch. Was quite impressive at the time.
kazinator 2y ago
The funny capitalization of moDernisT instantly gives away that it is an anagram of Tom's Diner.
kazinator 2y ago
> You can hear so many unnecessarily rejected sounds. That accusation requires evidence based in psychoacoustics. Just because you can hear it in isolation doesn't mean you can hear it if it is added back to the host audio. For instance when some quiet sound that is masked by immediately preceding loud sound is removed, of course you can hear that quiet sound in isolation! Your hearing has something like 120 decibel dynamic range, or better. You can hear differences in the compressed audio. Nobody can claims that there's no degradation in quality. Artifacts are obvious. Much more so at lower bit rates, though. MP3 starts to sound quite good around 192 kbps. The removal of those components is necessary. It is necessary to the algorithm so that it can achieve compression. Also there's this issue. If we take a signal and apply some modest EQ to it. Say we boost the bass and treble and cut me a little bit. Or any other EQ profile. If we then level match the two signals and subtract them from the other, there will be a difference: some aspects of the original material will be recognizably heard. For instance the difference between a slightly treble cut signal and the original will be the treble. But the trouble was not completely cut from the original. What you're hearing in the difference is not something that was entirely removed.

142 points by teleforce 2y ago

I would have liked a comparison with Ogg and perhaps other formats. I hear a lot about MP3 throwing away a lot more than Ogg but I'd love to see real data on it.

arch1t3cht 2y ago
ogg is a container format, not a coding format.
- wizzwizz4 2y ago
  Vorbis, then.
stavros 2y ago
I don't think the claim is that MP3 throws a lot more away, it's that Vorbis sounds better. The two are very different, Vorbis might be throwing a lot more away, but if it's only the stuff you can't hear, then it'll sound better.

Klaster_1 2y ago

The article doesn't mention at what bit-rate the difference track was made, anyone knows? Seems disingenuous and pro-"authentic" otherwise.

rob74 2y ago
If it's really the original MP3 version of Tom's Diner produced by the Fraunhofer engineers, probably not a very high bitrate. Aside from that, I would say that even back in 2015, MP3 was already on its way out and replaced by better (while still lossy) compression methods?
- porbelm 2y ago
  With space not much of an issue anymore, FLAC is pretty much the default nowadays. And even though Opus and AAC and others have better encoding than old MP3, but I guess a 128 Kbps MP3 encoded in the original Fraunhofer l3enc (the best back then) and one encoded with LAME will be different - and the LAME version will be "better" because of improvements in psychoacoustics? At least I remember l3enc being MUCH better than anything else at 128 Kbps (Xing lol, cymbal washing anyone?) before LAME came along.
  - klez 2y ago
    > With space not much of an issue anymore, FLAC is pretty much the default nowadays The default for what? Space is not the only consideration. What about bandwidth? I'm pretty sure spotify, deezer and the others are not transmitting FLACs, especially not at the base quality level.
Sesse__ 2y ago
It also doesn't really mention how this “lost” material was identified. If you just subtract the encoded from the original, then any phase difference will make it sound like material “disappeared”, while in reality, it just came very slightly earlier or later.
- rob74 2y ago
  I guess it was exactly as you write - but instead of slightly earlier or later, the "lost" sounds are the high frequencies (a lot of hissing, clicking etc.) - the actual sound is mostly still there, but slightly "muffled" because it contains only the lower-frequency components.
etwas 2y ago
It's 128kb. The useful and informative article is mentioned at the end: http://theghostinthemp3.com/theghostinthemp3.html http://theghostinthemp3.com/theghostinthemp3.html

0points 2y ago

This is why we dont encode mp3 in 96kbps or whatever.

CGamesPlay 2y ago

Found the original author's page about the project (no longer on the internet): https://web.archive.org/web/20211011015410/http://ryanmaguiremusic.com/theghostinthemp3.html https://web.archive.org/web/20211011015410/http://ryanmaguir... One interesting thing to note: this is a composition, not an analysis. It's not fully documented exactly what modifications to the "raw data" were made.

roelschroeven 2y ago
I can open the original link, https://www.theghostinthemp3.com/theghostinthemp3.html https://www.theghostinthemp3.com/theghostinthemp3.html, just fine. Maybe it's reinstated? Or there was a temporary problem?
- deleted 2y ago
  [deleted]

jonathanstrange 2y ago

So what. People listened to music on mechanical gramophones and enjoyed it. Too many audio engineers think it's all about the sound, when in the end it's about the music and the feelings it expresses.

a-french-anon 2y ago
You're naïve if you think that 1) Good enough prevents better from existing for some people, 2) Technological advances had the same consequences for all material. For example, orchestral music massively benefited from CD's increased dynamic range.
lukan 2y ago
"People listened to music on mechanical gramophones and enjoyed" Herman Hesse (or rather Harry Haller in Steppenwolf) used to get enraged by that - distorted garbage and a blasphemy to the godly componists. But then eventually he overcame it because of exactly that reasoning - if people enjoy it and are touched by the music - it works good enough. That is the purpose of music, not arbitrary perfectionism.
sevensor 2y ago
The way I think about recorded music is, the instrument being played is ultimately a set of speakers. That’s a musical experience to be encountered on its own terms. Live music is a wonderful thing, and presenting an illusion of live performance is certainly one thing a recording / reproduction system can do, but that doesn’t mean fidelity is the only thing we value when listening to a recording. Consider the punk music fans who enjoy a cheap stretched out cassette tape; I’m not going to deny them their enjoyment any more than I’ll deny the golden eared audiophile. They enjoy listening to different instruments. And this brings me to my point: the engineer is no mere technician whose job is only fidelity. The engineer is part of the performance through speakers, an artist no less than the musicians being recorded.
- mrob 2y ago
  Even the best stereo speakers have an inherent weakness: comb filtering between the channels. You can easily test this yourself by generating some mono pink noise in Audacity (or your favorite audio editor) and playing it on stereo speakers. Move your head and you will hear changes in the sound like a flanger effect. This is most obvious if you are close to the speakers and in an acoustically dry environment. Compare with the same sound hard panned to a single speaker. This is one reason why a real physical center channel improves clarity for movie dialogue.

a-french-anon 2y ago

This article could at least have a paragraph explaining (in a dumbed-down way) why various kinds of psychoacoustic masking (temporal, frequency) make what's removed almost inaudible anyway. Reading the linked source (https://www.theghostinthemp3.com/theghostinthemp3.html https://www.theghostinthemp3.com/theghostinthemp3.html), he at least used LAME, but at a fixed 128 kbps bitrate, not in VBR mode =( EDIT: nerds should read about sfb21 (https://wiki.hydrogenaud.io/index.php?title=LAME_Y_switch https://wiki.hydrogenaud.io/index.php?title=LAME_Y_switch), AAC, Vorbis and Opus (CELT) aren't just theoretical improvements

magicalhippo 2y ago
IIRC then 128kbps and lower means a 16kHz low-pass filter is applied, while it's not for higher bitrates. So in that case not just psychoacoustics.
- theandrewbailey 2y ago
  I was under the impression that MP3 can't represent any frequency above 16kHz, no matter the bitrate.
  - gwbas1c 2y ago
    Then they would all be 32khz, instead of 42.1khz or 48khz.
  - Sonic656 2y ago
    MP3 is only good at 18KHz at V0/320kbps, Yet AAC & Vorbis can handle full 22KHz at 256kbps VBR with hotter audio(-15 ~ -5db) their VBR models can just use 384 ~ 960kbps. Yes Vorbis can actually reach 1.35mbit since It has no frame cap.

Quarondeau 2y ago

Interesting approach. So are we only able to hear those sounds now because the rest of the music was removed, which would ordinarily mask the missing sounds? To say that the mp3-encoded version is not "what the artist recorded and wanted for us to hear" would imply that we can hear all sounds in the uncompressed recording.

varjag 2y ago
Yes exactly this. On a high bitrate stream these losses are imperceptible (yes even by you golden ears out here) due to how auditory filters work in human hearing.
- loa_in_ 2y ago
  That's assuming we understand human auditory filters AND human beings have uniform auditory filters all around
  - skrebbel 2y ago
    You can test this. Get a bunch of audiophiles in a room, play various recordings from various media (24KHz wav, mp3 at good bitrates and encoding settings, bad mp3, ogg, CD, vinyl etc) without showing them which is which, ask which one they like the most and see if the results correlate with the supposed quality of the source. I don't have a source ready but this has been a hot topic in audiophile land for decades and tldr is they'll pick out the really bad sources (eg <128kbps mp3) but not the rest. Basically the results look like those from a blind beer tasting test: no correlation between winner and supposed quality, except if the quality is especially bad. I'm no scientist, but to me "audiophiles who really care about this stuff can't pick out the good MP3 from the uncompressed original" is sufficient proof that MP3 is, actually, based on a sufficiently well-understood model of human hearing.
  - mrob 2y ago
    We don't need to understand those things to determine if the differences are audible. We can simply perform listening tests: https://en.wikipedia.org/wiki/ABX_test https://en.wikipedia.org/wiki/ABX_test A properly conducted ABX test is the most favorable condition possible for detecting a difference. If you can't ABX it you can't hear it. There's an ABX testing website with various lossy formats you could try: https://abx.digitalfeed.net/list.html https://abx.digitalfeed.net/list.html However, failing to ABX those specific samples does not guarantee you are unable to tell the difference in all circumstances. There are some sounds that are unusually difficult to encode ("killer samples"). This is an especially big problem for MP3. The LAME project has a collection of killer samples for MP3: https://lame.sourceforge.io/quality.php https://lame.sourceforge.io/quality.php More modern lossy formats are less susceptible to killer samples, but theoretically there could still be problematic cases.

NoPicklez 2y ago

Fairly lackluster article. Not all .mp3's are created equally and can vary in how lossy they are based on the bitrate. If you care enough to want to hear exactly what the artist wants you to hear, you just listen to the lossless version.

flanked-evergl 2y ago
The "lossless" version is in fact not entirely lossless. There is much that gets lost in the process of digitization, and even if there is no digitization happening the analogue equipment has some frequency response range.
- vinhcognito 2y ago
  Maybe theoretically but for our purposes doesn't the Nyquist–Shannon sampling theorem mean it's essentially lossless to our ears?
sumtechguy 2y ago
> Not all .mp3's are created equally No kidding. There are a decent number of options to pick for quality of mp3. VBR/CBR, bitrate, joint/stereo/mono, and so on. I personally just pick something that sounds fairly close to the original. But that only really matters for me when it is side by side. Give me a few days between picking and I can not really tell anymore.

Traubenfuchs 2y ago

...I think this person just created a new genre of music. Something like: "What's lost noise." I immensely enjoyed listening to the "lost material" of Tom's Diner and would like to hear more of this! Maybe one could diff with a lower quality version, one where more has been cut away, more is lost/left over? There are so many possibilities!

JKCalhoun 2y ago
Lost-Fi.
- Traubenfuchs 2y ago
  That's the Zeitgeist.

reliablereason 2y ago

You can try it yourself: ffmpeg -i original.wav -codec:a libmp3lame -b:a 192k output.mp3 && \ ffmpeg -i output.mp3 decoded.wav && \ ffmpeg -i original.wav -i decoded.wav -filter_complex "[1:a]aresample=async=1,volume=-1.0[inverted];[0:a][inverted]amix=inputs=2:weights=1 1" difference.wav

schroffl 2y ago
This produces an error in version 7.1 for me: [AVFilterGraph @ 0x6000008b59d0] More input link labels specified for filter 'aeval' than it has inputs: 2 > 1 [AVFilterGraph @ 0x6000008b59d0] Error linking filters Failed to set value '[0:a][1:a]aeval=val(0)-val(1):c=same' for option 'filter_complex': Invalid argument Error parsing global options: Invalid argument
- atoav 2y ago
  The theory behind it is simple: Subtract each audio sample in B from each audiosample in A. You can do the same thing in your DAW¹ by putting A (e.g. the original) onto one channel and B (the processed sound) onto another. Then you invert the phase of B and listen to/export the sum. This trick works also for audio gear that claims it does amazing things to your sound (here you just need to make sure to match the levels if they have been changed). Then you can look how much of the signal has truly been affected by your 1000 bucks silver speaker cable. ¹ Digital Audio Workstation, something as simple as Audacity should do the trick
  - TheRealDunkirk 2y ago
    > Then you can look how much of the signal has truly been affected by your 1000 bucks silver speaker cable. I have a friend who has spent ridiculous sums of money on audio gear. Like, he's in his 50's, and still lives with his parents (in part) because of it. Over the years, I've learned I will never convince him that he's being fleeced, but I've wanted to make a site to host such A/B comparisons for a very long time, to perhaps get through to others what a waste most of the "audiophile" gear is.

ezconnect 2y ago

When I first experience CD audio it was too high pitch compared to tape versions. MP3 came along and each song sound different depending on the MP3 compression settings.

jampekka 2y ago
Some CDs were mastered to have pre-emphasis that boosted high frequencies if the player didn't properly account for it. This caused them to sound "high pitched". https://wiki.hydrogenaud.io/index.php?title=Pre-emphasis https://wiki.hydrogenaud.io/index.php?title=Pre-emphasis

BurpyDave 2y ago

Ironically, the 'diff' is compressed anyway, because it's on Vimeo, so that's not the actual diff either!

taneq 2y ago
This is like those articles on 10-bit colour where they show you "how vibrant and rich" the higher bit depth is, but you're reading it on the same old 8-bit-per-channel monitor.

HPsquared 2y ago

This could also be done on visual compression with JPEGs. Or on video compression, for that matter. It just shows though that these diffs are invisible to a human - by design.

amelius 2y ago
Animals and aliens might cringe at these images and sounds, though. ps, you could do the same thing with watermarked content.

moomin 2y ago

I think the thing that's really sticks out is that the breath noise are gone, which is one of the things that gives the track its character. Willing to bet the same kind of thing happens to fret noise as well.

Fade_Dance 2y ago
My headphone rig was always optimized for a nice guitar sound, and I would second that. The sound of fingers on the string and the "pluckiness" of the guitar is what gets lost. Beyond that, the specific thing that i noticed gets lost is bass character on some tracks. Ex: Some drum and bass tracks just don't hit at low bitrate. This aspect sometimes feeds into the low guitar strings though, where they might have a bit less body. Lastly is of course sound staging, but that's something that a headphone setup is very sensitive to. As for quality differences, I basically fall in line with the consensus on this thread. FLAC and 320 are indistinguishable. 192khz is almost always indistinguishable and good enough, although there are some situations where it might be slightly noticeable. 128 is pretty easy to tell the difference with the good setup. There is also the rare track with amazing production and or very cool stuff happening somewhere in the spectrum, so I don't entirely write off someone wanting a plus one on the take above. "I absolutely love this jazz album. It's been a large part of my musical journey as a human, I can a/b test this at 320, but for this album, I really want it lossless." I can respect that. I got to know some of my test tracks pretty well (ex Black Sun Empire & Arrakis for bass), and while 192 was fine for other tracks, I wanted 320 or lossless on those.
- Optimal_Persona 2y ago
  As a musician recording ideas on a handheld Zoom recorder so I don't forget them, I switched from WAV to 320 MP3 a few years ago to save on file space and don't regret it.

no-such-address 2y ago

Funny article. "The exact master copy that the artist recorded and wanted for us to hear" In the digital era, does that even, uniquely, exist? "a set of standards created by a bunch of engineers in 1993" Nice! Was hoping the article would mention double blind studies about the ability to perceive differences and the quality between various audio file format, available elsewhere. Interesting, though not as overwrought as the reporting in this article.

chasil 2y ago
In fact, Glenn Gould's final recording of Bach's Goldberg Variations was rereleased from the analog tape recording, as the sound quality was better than digital of the time (1981). https://en.wikipedia.org/wiki/Bach:_The_Goldberg_Variations_(Glenn_Gould_album)#Re-releases https://en.wikipedia.org/wiki/Bach:_The_Goldberg_Variations_...

cladopa 2y ago

There is a trick there. A sound can mask another sound. You will not be able to tell the difference with both sounds playing at the same time, but if you subtract them you can hear it because there is no masking. I always loved to test the ears of my "Audiophile" friends. They will tell you how different MP3s are. You make a bet they can not differentiate them in 20 trials better than chance. I won with most people but some professional musicians that can identify little differences.

mrob 2y ago
>A sound can mask another sound. Details: https://en.wikipedia.org/wiki/Auditory_masking https://en.wikipedia.org/wiki/Auditory_masking Bernhard Seeber of the Audio Information Processing Group at the Technical University of Munich has some good demonstration videos on Youtube: https://www.youtube.com/watch?v=R9UZnMsm9o8 https://www.youtube.com/watch?v=R9UZnMsm9o8 https://www.youtube.com/watch?v=bU0_Kaj7cPk https://www.youtube.com/watch?v=bU0_Kaj7cPk
deleted 2y ago
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maep 2y ago
I spent half my professional career doing listening tests (MUSHRA and P.800), specifically on test items like Tom's diner. 128 kbps mp3 ist fairly easy to pick out, especially if you can compare it to the original. Double the bitrate and it's a real challenge. Modern codes like opus are much more efficient. At high bitrates they are fully transparent and anybody who claims to be able to hear a difference is full of shit. Put them in a controlled setting and they fail every time.
- anotherhue 2y ago
  LAME @192kb/s VBR was transparent 20 years ago, that said FLAC is still a good choice because storage is cheap now and you don't want to have to deal with a copy-of-a-copy situation. Some Young folk think that 24bit/192kHz is the one-true-form who would think a 16/44 FLAC is a lossy encode, and then there's the vinyl folks. (I like vinyl, but not for the fidelity). Required reading: https://xiph.org/video/vid2.shtml https://xiph.org/video/vid2.shtml

grishka 2y ago

Now I want this comparison for Opus. It doesn't do that whole psychoacoustics thing, does it? But it also somehow manages to ~double the compression ratio compared to MP3 without any noticeable difference in the sound quality.

bjoli 2y ago
I haven't gotten into reading about audio and video compression yet, but it from where I am standing now it really looks like magic.

gwbas1c 2y ago

In 1999 when MP3 was getting attention, I tried to do this. I encoded a file, then inverted it, and mixed it back into the original. It didn't cancel anything out. The reason: Mp3 dramatically alters phase. Because all the phases are different, it's hard to naively determine how the signal is altered. Years later, I took the time to write a series of tools to investigate lossy audio: https://andrewrondeau.com/blog/2016/07/deconstructing-lossy-audio-the-case-for-lossless https://andrewrondeau.com/blog/2016/07/deconstructing-lossy-...

tecleandor 2y ago
Oh! Is that an artifact of Joint Stereo encoding ?
- gwbas1c 2y ago
  > Oh! Is that an artifact of Joint Stereo encoding ? I personally don't understand enough of MP3's internals to explain that. What I assume is that, because MP3 internally stores Fourier transforms in the frequency domain, instead of the time domain, it uses very few bits to store phase. This will result in phase shifts. Hopefully, someone else can give a better explanation of this than I can. Basically, think of it this way: 1: Imagine a frame of 64 16-bit samples. (1024 bits) 2: Than its converted to 32-bit floats: 64 32-bit samples. 3: Than its transformed 4: Now there are: 1 DC bias 32-bit float, where phase is 0, and 32 frequency amplitude floats, and 32 phase floats. (The other 31 frequency/phase pairs are duplicates and don't need to be stored.) These floats need to be quantized so that there are many less bits. Some of it can happen by using very few bits to store phase, and some of it can happen by using less bits to store amplitudes. Again, someone else can probably explain this better than I can.
  - tecleandor 2y ago
    Interesting. In a quick internet search seems like people finds a small phase shift on MP3 enconding, but I haven't found details on why. Haven't had much time to find it, though...

sdk77 2y ago

Very interesting! The audio of Tom's Dinner rejected by the encoding sounds mesmerizing to me. I still find it to be musical - it reminds me of a record I bought a really long time ago, it was called modulation & transformation on mille plateaux, it's a collection of songs in the abstract and experimental genre.

jonnycomputer 2y ago

"What MaGuire has proved here is that the songs we listen to every single day are not the exact master copy that the artist recorded and wanted for us to hear. Instead, they are slightly stripped versions of their art run through a set of standards created by a bunch of engineers in 1993. For many people, that won’t matter. The songs sound almost the same, but the compression of music into an MP3 format is an important question to weigh when considering artistic intent and analyzing songs that aren’t exactly the original." I feel like this analysis isn't well grounded in what artists and sound engineers actually do, or how they think.

chasil 2y ago
When I've used lame --preset extreme, it uses variable bit rate which was developed some time later.
Optimal_Persona 2y ago
I honestly didn't know that was a matter of contention at this point...it's very common for pro (or smart amateur) audio engineers to actually listen to the effect of different digital compression algorithms and adapt accordingly. Things like Sonnox Fraunhofer plugin let you do this. https://sonnox.com/products/oxford-fraunhofer-pro-codec https://sonnox.com/products/oxford-fraunhofer-pro-codec - In this month's TapeOp magazine, Jeff Jones mentions he always monitors through such a codec so he can adjust accordingly for the best sound regardless of final format https://tapeop.com/interviews/163/jeff-jones/ https://tapeop.com/interviews/163/jeff-jones/ - IMO Rush's Moving Pictures is one of the best (and best-sounding) albums from the '80s despite the fact that it was mastered to a Sony digital device that's renowned as terrible sounding, with only 14 bits of usable resolution. How/Why? The production team monitored through the Sony system while mastering and made tweaks as they went to account for its limitations. - I've had my own music professionally mastered at high resolution, and discovered that converting hi-res to MP3 (even at highest bitrate) caused digital peaks over 0.0 dB, I fixed by normalizing to -0.3 dB. TBH if the resulting file is going to be transmitted over BlueTooth it's going to be further degraded. And yet people can still make a strong emotional connection to the underlying music and artist...

pvillano 2y ago

Two instances where lossy compression failed for me are the movie Koyaanisqatsi and songs by the artist TOBACCO. Koyaanisqatsi has a lot of film grain and TOBACCO uses a lot of distortion. There is noise in there, but it's very deeply mixed into the signal.

ipunchghosts 2y ago

There's still audio motifs in there that can be further optized out. If the remaining audio was noise like, I would say we reached the compression limit.

Agraillo 2y ago

Why Tom's Dinner? Because it is a cappella. There's a book "How Music Got Free" by Stephen Witt [1] detailing the history of mp3 format and related events. It is a very good read and there's an explanation Increases in processing power spurred progress. Within a year Brandenburg’s algorithm was handling a wide variety of recorded music... But one audio source was proving intractable: what Grill, with his imperfect command of English, called “the lonely voice.” (He meant “lone.”) Human speech could not, in isolation, be psychoacoustically masked. Nor could you use Huffman’s pattern recognition approach—the essence of speech was its dynamic nature, its plosives and sibilants and glottal stops. Brandenburg’s shrinking algorithm could handle symphonies, guitar solos, cannons, even “Oye Mi Canto,” but it still couldn’t handle a newscast. Stuck, Brandenburg isolated samples of “lonely” voices. The first was a recording of a difficult German dialect that had plagued audio engineers for years. The second was a snippet of Suzanne Vega singing the opening bars of “Tom’s Diner,” her 1987 radio hit. [1] https://en.wikipedia.org/wiki/How_Music_Got_Free https://en.wikipedia.org/wiki/How_Music_Got_Free

chrsgrrtt 2y ago

I developed a streaming service many years ago; Dolby wanted us to use their codec for the audio, and they used a track just like this as the primary basis of their sales pitch. Was quite impressive at the time.

kazinator 2y ago

The funny capitalization of moDernisT instantly gives away that it is an anagram of Tom's Diner.

kazinator 2y ago

> You can hear so many unnecessarily rejected sounds. That accusation requires evidence based in psychoacoustics. Just because you can hear it in isolation doesn't mean you can hear it if it is added back to the host audio. For instance when some quiet sound that is masked by immediately preceding loud sound is removed, of course you can hear that quiet sound in isolation! Your hearing has something like 120 decibel dynamic range, or better. You can hear differences in the compressed audio. Nobody can claims that there's no degradation in quality. Artifacts are obvious. Much more so at lower bit rates, though. MP3 starts to sound quite good around 192 kbps. The removal of those components is necessary. It is necessary to the algorithm so that it can achieve compression. Also there's this issue. If we take a signal and apply some modest EQ to it. Say we boost the bass and treble and cut me a little bit. Or any other EQ profile. If we then level match the two signals and subtract them from the other, there will be a difference: some aspects of the original material will be recognizably heard. For instance the difference between a slightly treble cut signal and the original will be the treble. But the trouble was not completely cut from the original. What you're hearing in the difference is not something that was entirely removed.