When I was at Berklee [0] we had considerable ear frequency response training that would involve weekly hearing quizzes where they would boost/buck certain frequency bands of either pink noise or music and you had to pick out which band was made louder or softer. Additionally, we had a quiz where you had to listen to lossless vs mp3 at different bit rates and determine which was which. We also did neat things like trying to pick out what type of guitar and pickups someone was using in a recording simply by ear.
320 kbps mp3 vs lossless 16bit WAV (cd quality) is extremely difficult to hear the delta without training, but if you know what the source instruments are supposed to sound like, such as hats and you can focus on their frequency band, you can hear compression artifacts and comb filtering.
This course really changed how I take in music and sounds in general.
I was hoping the course you recommend would be available for the general public, but at >$1000, this is a no go. I would happily pay for something like this, but not this much.
Can anyone recommend an affordable equivalent?
Harman Audio created a free "How to Listen" application years ago for both Mac and Windows, which "consists of a number of training exercises where different kinds of timbral, spatial and dynamic distortions commonly found within the recording and audio chains are simulated and added to music." It isn't about MP3 compression specifically, IIRC, but it's a good critical listening course.
But: while it's still available, it's about ten years old, and I have no idea if it still runs on modern hardware.
In my experience I will hear a shimmering effect that sounds like a tremolo where the volume of the hi-hats and other cymbals are fluctuating. It also causes the overall sound of cymbal crashes to be mushier and less distinct. In addition I've noticed an effect where certain sounds such as the attack of high-gain guitar is stretched and doesn't sound consistent throughout a song. It seems like these artifacts are most noticeable in higher frequency ranges or perhaps transients.
Audio has its own equivalent. It's not noise, it's more of a gargly quality. You can hear it very clearly at more extreme compression settings.
The overtone movements become disassociated and start splitting up into disconnected sine wave blocks. Transients also become softened.
With MP3s this is obvious at the high end, while lows/mids sound passable. The compression algorithm has more frequency bins at the high end, so the damage is more obvious there.
There's also a loss of low-level detail. Reverb tails lose definition, and the music loses front-to-back depth.
There's something about the really low tones that simply... changes at lower bit-rates. High-frequency tones are somehow clearer when uncompressed. It feels like the tones are right, but the waveforms distorted. Your rippling lake MPEG analogy is pretty much spot on.
That being said, age, tinnitus and amazing progress within the audio compression sciences have made it more difficult for me to separate uncompressed WAV and 320 kbps MP3, though the 320 kbps files in the article had noticable artifacts in most cases.
Not OP, but just a random ex-audio guy who did his fair share of listening experiments with different formats and bitrates.
FWIW, it had the complete opposite effect on me. I stopped caring about things like flac or other lossless formats for general music listening.
If the only way to maybe sometimes pick out a 320kbps mp3 from a flac one was to be on my best headphones, in my quietest room, focusing all my attention... the difference doesn't matter for practical listening. My ears may just be garbage, but it made me skeptical of people who make bold claims about what minutia they can discern -- although magical claims are more prevalent in the audiophile world ("these cables sounds great!") than the actual audio production world.
Not an audio professional but I'm the sound guy for a nonprofit.
Along similar lines, I can't tell the difference between a $250 microphone and a crazy expensive mic in anything but ideal conditions -- and even then, I need to hear them side by side.
+1; there are so many more important factors before the smaller things start making a very noticeable difference. A Neumann U 87 into a 1084 or VoxBox will sound marvelous with good talent giving a great performance, but that performance would still sound damn good with a $100 Shure SM57/58 into a Focusrite interface costing about the same. It’s never been easier or cheaper to be able to make music that’s good enough to get plays on Spotify.
Buying a Ferrari won’t make you Michael Schumacher, but he could probably lap your car round a track faster than you ever thought possible.
>does it ruin your appreciation for great music in lower quality formats?
As someone that has high-end gear, I can absolutely tell the difference in most cases between 320kbps MP3 and lossless when switching between the two versions of the same song. It's hard to put a finger on, but it just "feels" different to my ears/perception. However, after listening to exclusively one or the other for a few minutes or not A/B switching the same song, I have a much harder time detecting the difference.
Quality of the master and quality of the equipment matter much more, IMHO.
+1, I think for those with even a little musical background, the brain fills in the details to a much greater extent, especially if it’s something you’ve heard before on a good signal chain / live. You know what’s there, so you know what to listen for, and your brain kinda just interpolates the rest.
This is not my idea btw, I read it somewhere, just forgot where I read it. Googling a bit around “musician audio quality ‘fill in the gaps’” didn’t ring any bells. I’ve been reading up on acoustics and audio engineering over the past year trying to get my personal music project together, so it may have just been from a book as well.
Not at all, but I do have much greater appreciation for the time that sound engineers took to cram as much frequency information into the fixed size spectrum box using level variation, panning, parametric equalizing, and compression. All of those things we had to listen for and critically evaluate in that class.
Let me put it this way: I prefer Music-A so much more than Music-B, that I'd choose Music-A playing on an AM-radio over Music-B playing on an FM radio.
Fidelity is indeed swell, but how important is it really? A lot of the music I like best, I first heard on a crappy recording or a shitty sound-system. There were A LOT of hit songs recorded before FM came along, and yet that 10k AM bandwidth (e.g. when the Beatles were big) didn't keep people from buying copies by the millions. Gasp! How can that be?
How much of the time I'm actually listening to music I like do I actually care about the fidelity? If band A's crappy cover is available in FLAC, will I choose it because band B's brilliant cover is only 128K mp3? NO.
This eternal argument has little to do with music. I still like 'Louie Louie' by the Kingsmen (a recording that cost $36), never mind how shitty the studio was. I'll still choose it over a FLAC from the studio master of "Never Gonna Give You Up". I can hear around the 'flawed' repro. Two kinds of 'purism' at work here. Pure genius beats pure fidelity.
You're right, but I think you're missing that it's sometimes genre dependent. The Beatles material was crafted to make it sound good over AM radio, and it does. I'm also a fan of loads of electronic music but as far as I'm concerned there's no point listening to the latter on a system that can't reproduce the bass. (Unless it's earlier material like the Prodigy which again I think was crafted to sound good on crappy blown car speakers - exception that proves the rule!)
That’s my position, too. I listen to a lot of jazz recorded in the 1930s and 1940s, sometimes with audible surface noise from digitized 78s [1], and the musicianship comes through just fine.
I can read a novel or poem on a smartphone, in a paperback, on a fourth-generation photocopy, or in fine letterpress printing on 100% cotton paper. The reading experience might be a bit more comfortable in the last case, but the author’s words are the same in all.
I like the Kingsmen’s “Louie Louie,” too. I played it for my grandson yesterday.
Personally I want to listen to the music I like in the best quality available.
Call me a snob, but usually I'd rather listen to a totally different Band C's song in at least 320k mp3 (preferably FLAC) than either Band A or B's in less quality.
But I enjoy music on a technical level as well. I find it very difficult to enjoy bad mastering or audio quality. There is enough well-mastered and recorded music out there nowadays that I don't feel I need to be subjected to badly-recorded music.
There are some recordings where I really wish they were done better, but very few that are really badly recorded that I can enjoy, no matter how good the music itself might be.
I'll take what I can get, basically. If I don't have access to good gear or good recording/compression quality, I'll use what's available. But if there's a song I like and I have the option of using good gear and lossless audio, I personally find it can optimize my enjoyment of the song.
I picked 4 out of 6 and never picked 128kbps. I am pretty sure I can reliably weed out 128Kbps, but WAV vs 320Kbps is more or less luck. Jay-Z was the easiest as the chiptune sample is a rectangular wave and hence has a very high frequency component that actually makes sound less palatable, not more. "Hissiness" of strings helps on classical and Neil Young, but pop music (Katy Perry and Coldplay) with its flat wall of sound is much harder - I was listening for "z"s and "s"s and high hats. Acapella (Suzanne Vega) was the hardest for me and I think I just lucked out.
Suzanne Vega ought to be the hardest thing in the world to find an MP3 artifact in, since "Tom's Diner" was the primary litmus test used in the development of the psychoacoustical model.
I'd argue the opposite, "voice is the hardest to get right so we chose a vocal track" -> voice is hard to get right, so they got it least right..
For me, Tom's Diner was the easiest of the lot to pick out.
I did the quiz once and also got 4/6. I alternated through all the samples around 4 times before making each choice. I messed up on the only two where I felt baffled by the choices, so at least I know what I don't know. :-)
I chose the 320 kb/s for Katy Perry but was vacillating between that and the actual WAV. The one I completely screwed up was Jay-Z where I picked 128 kb/s. However, I knew I was lost with so much electronic glitching and distortion. While I heard differences, I had no idea what was a compression artifact versus a synthesis/mixing artifact. I also had no prior listening experience for either of these two, so no preconception for how it should sound.
For the other recordings, I think I could hear the masking of the psychoaccoustic models. Essentially, in the WAV I heard a "fuller" mix with overlapping sounds, while in the compressed versions I heard dropouts where a dominant sound was present in what felt like isolation. For Suzanne Vega this wasn't so easy, but I felt her voice and the reverb were more clear and natural in the WAV.
This was just with my old Sony MDR-NC22 noise canceling ear buds driven by my Thinkpad audio jack. My idea of a hi-fi environment is a Yamaha receiver and reasonable tower speakers like my Polks.
I have a similar experience with being easily able to distinguish between uncompressed and 320kbps MP3s with certain kinds of music. Music with a lot of emotion in the singer’s voice or with both instruments and singers spread over a large area (Bollywood music from the late 90s) are easier to distinguish. Of course, I tested this with two different streaming services, so it may just be differently processed files. Modern pop songs on the other hand sounds the same to me irrespective of the level of compression.
In addition to headphones, and hearing, there is also training.
There are a number of audio artifacts in MP3s that are much more noticeable after training. However, many of them become much less discernable with variable compression rates and lower compression (>256kbps). It's possible to train on those artifacts, but I've always avoided it since I don't want the "projectionist effect" of making standard recordings less enjoyable.
This is not the case for many audiophiles. They want to fix all the bugs!
Projectionists needed to learn to spot the little white blobs in the corner of films so that they could switch reels. Once trained, you can basically never miss it and it will annoy you every time you see it. Also, if you develop visual compression schemes (e.g. MPEG, H264, etc) you will basically always see the blocking artifacts and chroma aberration in compressed video. That leaves you hating all DVDs, blueray, and digitally compressed video since looking at it becomes work.
>you will basically always see the blocking artifacts and chroma aberration in compressed video
Netflix is AMAZINGLY low quality, even on the higher-quality tiers, and working with compressed video was one of the factors that ultimately led to me cancelling my subscription - I'd rather pay (or yarr) for an actual copy rather than something that's almost more compression-artifacts than actual video.
Not to mention TV post-digitalization. Nowadays one can't watch TV or streamed videos without getting annoyed over how poor trees, bodies of water or audiences in arenas look due to the compression artefacts. Ironically, I prefer buying a DVD for a few € to streaming a movie simply because there's less visible artefacts.
DVDs are horrible quality though. Compression everywhere and low resolution. The streaming services that I've watched are all better than DVD, at least 720p.
If compared to DVB-T with far to low bitrates, or streaming over crappy Internet connections, DVD's are far better in comparison.
Blu-ray is obviously the better option, but they're more expansive and not always available, especially when it comes to stuff that isn't produced in the U.S.
Similarly, for sound effects, once you learn what the Wilhelm Scream is (https://en.wikipedia.org/wiki/Wilhelm_scream), you start noticing it everywhere it's used, and it can pull you out of the immersion in the film you are watching.
I guess in a sense, clipping the waveform essentially allows you to keep dynamic range. To avoid the clipping (but to keep the gain on the quieter portions) normally one would have to compress it (more than it is / to the point where it would just graze the limits, but not clip).
The cost, of course, is the the clip is horribly audible in the final waveform.
That and the Katy Perry were bad choices, they were mixed to sound like shit. The breakdown in the Jay-Z track was really a good place to look for a difference.
Just to add to stats (which really, how useful is it with the variables of audio setup and subjective preference): 5/6 with Mobvoi TicPods using aptX HD. My wrong answer is the Jay-Z one and I picked 320kbps.
I listened to the samples multiple times for each test. As I went through, I kind of used a strategy to skip to and compare the busiest parts of the songs for "muddiness". Still, it was always very tough to tell between 320kbps and WAV. Most of them were decided on a subjective "it just sounds slightly clearer and fuller". I think I had the easiest time with the vocals only track; can attribute that to being a vocally-focused listener.
Side note on that last point: Only recently have I noticed how little I heard of drums in my music listening. Picked up edrums recently and realized how lost I was at drum beats "ideas". For melodies I feel like I have a large library of ideas to draw from. On the flipside, now I notice drums a lot more. Hopeful that others can relate to this too?
I did a similar experiment to myself years ago with a MiniDisc player and found that 128kbps was indeed the sweet spot for myself. Especially when space mattered.
While I am sure that more expensive equipment or years of dedicated study can help me discern the difference, I don't feel like my lack of refinement hampers my enjoyment of music in any way. If anything, I have no interest in training myself how to notice peas under mattresses.
I am glad there are people who master their medium; but I can't be a master of all media and the intricacies of their particular language, history and references -merely to consume them.
Live and let live. I'd like less judgement for being like The Matrix character Cypher:
“I know this steak doesn't exist. I know that when I put it in my mouth, the Matrix is telling my brain that it is juicy and delicious. After nine years, you know what I realize? Ignorance is bliss.”
I got 4/6, missing on the last two. Interestingly, I picked 128kbps for those, and they were more "pop"-y songs. Maybe I'm trained to like some more compression on pop songs?
I was also listening on my work setup, which isn't ideal: Bose QC35s plugged into an O2 amp and D50 DAC. I'd like to think I would've gotten more right with some flatter headphones.
The main things I noticed when comparing were:
* low end was smoother on the bass
* clarity of high end hits ("pss/tss" sounds)
* some vague sense of better imaging, like the sound stage was better
I didn't notice much difference in the midrange or vocals, but that might be because of the gigantic V-shape of my headphones.
This kind of test is really tricky because you are either trained to recognize the artifacts of MP3 compression, or you are not in which case you just "trust your gut". And if you are used to listen to compressed music (e.g. youtube), you will probably pick that.
And that is not even taking into account that MP3 (despite being really old and not the most efficient at this point) is good at what it does, especially at 320k.
Usually what newer codecs buy you is the same quality for lower bitrates. AAC-HE, for example, would be pretty hard to pick out at 128K. Opus at 96K. Push either of those to 320K and I doubt anyone can really tell the difference.
A similar thing happens with MPEG2. At a certain point, all the bits thrown at it make it indistinguishable from source. Where newer codec shine is when you start saying "what if we do this at 1000kbps" or "600kbps".
This makes me wonder—these discussions usually assume that compression artifacts make audio sound worse, but is that always necessarily the case? Could they just as easily make it more pleasing?
“Whatever you now find weird, ugly, uncomfortable and nasty about a new medium will surely become its signature. CD distortion, the jitteriness of digital video, the crap sound of 8-bit - all of these will be cherished and emulated as soon as they can be avoided. It’s the sound of failure: so much modern art is the sound of things going out of control, of a medium pushing to its limits and breaking apart. The distorted guitar sound is the sound of something too loud for the medium supposed to carry it. The blues singer with the cracked voice is the sound of an emotional cry too powerful for the throat that releases it. The excitement of grainy film, of bleached-out black and white, is the excitement of witnessing events too momentous for the medium assigned to record them.”
Personally I've had a similar experience: hearing a song somewhere played on low-quality speakers and liking it, then later finding the high-quality version and not liking it as much.
That's exactly how I feel about 4k TV or 48 or 60fps films. Unless it's sports or certain modern/comics action 720p with color depth seems about right. But as the Brian Eno quote says, it's what I'm used to or learned to like.
I remember I had one friend that when CDs came out didn't like them, but found that if he recorded them onto S-VHS tape sounded best of all, so he ended up using S-VHS like cassettes at home. In theory could have got a portable but that's taking it a step too far.
I suppose it's nothing surprising, but I've had the opposite experience, hearing Michael Jackson's Thriller from SACD on highend speakers after listening to it from tape in my youth. *chefs kiss*
They certainly can, it really depends on the music being compressed and the user’s preferences. MP3 often sounds "cleaner" thanks to shaving off part of the spectrum that is not as audible.
I imagine that by now, skilled audio engineers who are working on distribution tracks for artists will well and truly have the hang of mixing for the various lossy codecs out there.
Sometimes it may be helpful to look at some fundamentals from a completely analog point of view before trying to make the most out of what digital has to offer.
If so Chapter 14 Fidelity and Distortion in Radiotron (RDH4) can be a good place to start:
>The values of total harmonic distortion to provide objectionable distortion are 2 % with a frequency range of 15000 c/s and 10.8% with a frequency range of 3750 c/s for music, and 3% and 12.8% respectively for speech, with a pentode.
Even if almost nobody is using pentodes any more, you could do worse.
The whole thing is 1400 pages, you can see what engineers were up to with their slide rules back then.
Chapter 19 if you really want to know about decibels . . .
For every case, I picked the one I feel subjectively the best. Interestingly, I still prefer the 128k version from the "Tom’s Diner" example, even after knowing the "correct answer". This is amusing because according to the descriptions, this song was chosen as the "benchmark" for the algorithm, which could have resulted in some overfitting ;)
I've also noticed this with some DJ tracks that I play. I have some old ones that are 128kbps and I prefer a few of them over the higher bitrates. Gives it a lofi feel, as is the case.
Wow I kept picking the 128kbps MP3. Im using Sennheiser HD58X. I'm plugged into the headphone jack of my Desktop monitor which itself is getting audio via USB-C. Maybe the amp is bad in the monitor? Wonder if an external DAC would be any better.
If you keep picking the 128kbps track it does mean that you can hear a difference, and maybe that you subjectively prefer the more compressed audio.
That's why usually good audio tests use ABX where it's not about seeing if you can identify what is best, but rather if you can identify a difference at all. See for instance: http://abx.digitalfeed.net/
EDIT: I actually just tried the test and... ended up doing the same thing you did. On the Neil Young track I could hear a definite difference in the high pitched violin-like background, but I ended up selecting the 128kbps track as the "highest quality", maybe because the compression made it sound smoother.
The HD58X uses 150-ohm drivers. These can be used without an amplifier just fine, but detail will be lost and they'll be quieter. My general rule is anything 80ohm and over will want an amp... You DEFINITELY want an external DAC+amp for those 58X's.
The DAC chip in a monitor will be the absolute cheapest one they can source, and is probably receiving a lot of noise from the monitor itself as well. The good news is that you don't need to break the bank to get something decent; a budget of ~$100 will get you something a million times better than a screen could ever deliver.
I'm going to confine myself to DAC+ Amp devices here. The main trick to suss out audiophile-snakeoil is checking the specs to see what the Total Harmonic Distortion is when the volume is maxed out (ie, before the signal starts clipping). THD is easy to have low (<0.01) when there's little load. With amplifiers of all kinds, quality is determined by how low the THD is when it's turned up. If it's >1% then it's poo, if it's ~0.1% then it's okay, if it's 0.01% or less then it's pretty good.
The $400 Element II [1] from JDS Labs (makers of the legendary Objective-2, a DAC+Amp that was designed to give top-end performance at bottom-end prices) is very literally as good as it gets without snakeoil entering the equation. THD when under a 150-ohm load is 0.0008%. That's very good. This is what I'll be upgrading to if my setup at work ever dies.
My work rig is an Aune T1 [2] which retails for $250 but Massdrop does them for $100-$150 sometimes. Yeah it's a "tube" but that doesn't matter as much as myth would have you; today's tubes are pretty transparent unless you deliberately get something that isn't. THD is good however nowhere near JDS Labs-level of perfection.
The Fiio Q2 mk2 [3] can be found for as little as $100 and it's a solid performer, though I haven't personally used it, Fiio's reputation is impeccable.
When I was in the mark for a decent DAC last year, I settled on JDS Atom DAC. Cost $100 I believe and seems like their support is good. Though I don't have anything else to compare, it seems good so far.
I think in this case it's more like you've bought a truck but you put a lawnmower engine in it.
It's a big step between esoterics and "please don't use the cheap shitty DAC that was integrated into your monitor to satisfy a feature checklist".
The amp part will matter a lot the more impedance the cans have, and many of the better headphones are high(er)-impedance. Might as well spring for a headphone amp + DAC combo just to see how it sounds. It's not that expensive, FiiO has some for under 80 bucks, for example.
I do. I'm very pleased with the (relatively) flat response using powered studio monitors in my home theatre setup, which are (relatively) affordable compared to something with similar performance purchased from a hi-fi specialist.
So, this is a combination of 6 tests, and each has 3 variants of the answer. That's definitely not enough to make useful statistical conclusions about the ability to detect bad 128kbps audio, simply because it is rare.
Indeed, the probability of choosing all the best clips by chance is 0.13%, and the probability of never choosing the worst (also by chance) is 8.8%. Choosing almost all the best clips, with exactly one mistake, would have the probability of 1.6%, and choosing exactly one 128k mp3 has a 26% chance.
Disclaimer: I have not actually performed the test.
I could easily tell the difference between 128kbps and the other two. Telling the difference between the 320kbps and uncompressed was close to random at first. Once I know which is which I can pick each out. So the problem isn't being able to tell that there's a difference, but telling which is closer to the original. Each piece can have a clear tell, like the timbre of a particular note that's played in a busy soundscape. I used to have excellent hearing and now my high frequency hearing drops off at about 15kHz. I think because of this sometimes content that has already had some low-pass filtering may seem more natural where I can hear all that is being reproduced. I could tell the differences using an LG G8 QuadDAC on JVC HA-FXD80 earphones or the Surface Go speakers 3/4 volume at arms length.
[Edit: I'd also been drinking tonight which raises my blood pressure and I can hear slight ringing in my ears triggered by bloodflow or viscosity change. I can try the test again tomorrow.]
The best indicator for sound qualify is how long I enjoy actively listening to music. With a good source and sound production I can literally do nothing other than listen to the music for hours and not fidget with or read anything. Same with club systems. If it's not good I won't really have a good time for long. The Alpha Dynacord at Stereo in Montreal was amazing the time I heard it with 45 vinyls being played.
Used $50 Audio Technica ATH-M20X on $90 Behringer UMC202HD.
Did audio production in the past (amateur level).
Failed Coldplay, and Neil Young, the latter I argue has such poor source quality that it's hard to tell a difference.
Tip: listen to the "air" of you want to spot the difference, that's the first stuff the compressors throw away. "Air" is the ambient sound of the room, the tail of the reverb, the stuff you hear when you don't hear anything if that makes sense.
6 for 6 listening directly from laptop speakers. That being said I really had to listen intently to discern any difference between 320 kbps vs uncompressed. Everything sounds ever so slightly flat or muffled.
Oh, how well can I hear the difference in quality, from my laptop speakers? Geez, I'd have a better chance on my phone, and that would be no chance at all.
I listen to Tidal through a USB DAC connected to a preamp -> amp -> respectable speakers.
And I can definitely tell the difference. But no one can tell the difference on laptop speakers, the idea is nonsensical, or worse, spreading the idea there is no difference between uncompressed audio and highly compressed audio.
> But no one can tell the difference on laptop speakers, the idea is nonsensical, or worse, spreading the idea there is no difference between uncompressed audio and highly compressed audio.
That is the point, isn't it? The absolute vast majority in of people listen to music on anything but gold-plated pre-warmed high-end systems. They listen to music on phones, on portable speakers, in headphones, while driving, cooking, taking care of kids, working... For any practical purpose there is no difference between uncompressed and compressed audio.
6/6 on poor laptop speakers, but based on a side-channel - the one that takes a second to load is the WAV. Actually listening, it's tough to make out 128kbps MP3 vs WAV on this setup - there's a slight difference, but only when listening closely and directly comparing the two samples.
Generally I can usually (depending on the song) pick out 320kbps MP3 vs WAV on reasonable headphones (ATH-M50x), but I don't have a pair to hand to test these samples.
The only one I got dead wrong was Coldplay. That also happened to be the furthest from my usual listening habits, I don't know if that had any effect.
For the others I picked at random one of the two that sounded the most similar. In no instance could I tell 320Kbps and WAV apart. Could have been just luck that I didn't pick 128Kbps more than once.
Hardware is Bose QC II on low noise cancellation, connected by wire to a MacBook Pro and ears, one of which can hear crickets, the other not.
5 out of 6, and the wrong one was 320 kbit, but that one was very close between them.. between wav and 320 mp3, it's difficult enough that I can't put firm words on it.. the wav is "more" or "larger" or "fuller" without there being anything specific I can point at.
5 out of 6 as well using a cheap pair of Koss KSC75 with some late 30's ears. It was the Katie Perry Song that I missed. "larger and fuller" is a good way to describe it.
The inclusion of the Neil Young got a smirk from me because of his past antics with promoting his “Pono” high-res music service. (2012-2017)
The Jay-Z is there for similar reasons, explained in the opening: he was promoting his own high-res music service. (Still theoretically extant but he’s gone back to streaming on Spotify.)
And the Suzanne Vega track is there because it was one of the three or four songs that was used as a benchmark for the development of the MP3 format; if there is any song that should sound good as a shitty low-res mp3, it is Tom’s Diner.
But what about the Coldplay? And the Katy Perry? And the Mozart - and maybe this particular recording of Mozart? Are any of these entwined with the history of lossy audio compression, or attempts to sell lossless compression to a wider market than audiophiles?
Can't say I did well on the test. Actually made a decision on 2 of them and got the 128kbps on one and WAV on the Suzanne Vega (which must have been luck). The rest I just clicked something to read the snippets.
And I've got HD598s and was listening through a UMC1820 interface.
That said, I'll take lossless FLAC files for a download any day, simply because it's future-proof in a way compressed audio isn't. I'm quite happy to listen at 320kbps on Spotify most of the time, or transcode to 96kbps Opus any FLAC I have when I load it on my phone.
One thing to consider is that this for listening. However as soon as the audio needs to be manipulated e.g. use it in a film, sample it, EQ it heavily, adjust the dynamics DJ with it, low quality audio (especially mp3) gets exponentially more noticeable.
In the end that compressed great song will still beat the uncompressed lame one, but how noticeable compression gets also depends on what you do with the material.
Music is engineered to sound good on every quality, up until the point when it is heard on low quality devices and adjusted so it is still perfect. It causes finding differences so difficult. It is not only that compression is so good, or that we don't hear the difference. It is also about the music being made in a way we shouldn't hear it.
This test result is a tribute to sound engineers who make music good on every device.
So despite a lot of comments here saying that can't tell the difference, there are still people out there determined to pay more money for Tidal's "HiFi" sound quality and with both Amazon and Spotify (Apple also?) also now coming out with "lossless" quality options. It may be considered snake oil by some but snake oil sells (not news).
Happily surprised I managed 5/6 on the sony xm1000m3 (man I hate their naming) over bluetooth. Didn't expect that!
(my audiphile dad did 'train' me... made me listen to the highend stuff he's been building/crafting since I was little. I owe him disliking 128kbps on almost every setup ]: I'm fine with 320kbps for most things though)
From what I recall about compression, the artifacts are at the high frequencies which also tend to get muted with age / sound exposure. Rather than play the game of determining if you can hear the difference, are there any diagnostics that can work backwards and determine the max bit-rate one could benefit from when listening?
Interesting, whenever I failed it was between 320kbps and uncompressed wav. I know that I did a blind test about 10 years ago and could distinguish between ~192kbps variable bitrate and lossless audio with my left ear, mostly influenced by percussion (left ear for some reason can hear 21kHz frequencies).
Uncompressed CD audio is 1.4 Mbps. FLAC can do roughly 2:1 compression, so that's 700 kbps. It's not crazy that modern mp3s (encoders have gotten better over the years) sound good at half the size of something lossless. They're only removing subtle details at that point.
You can see the same thing with high-quality JPEGs vs PNGs for photos. There's a decent size reduction, but to see the difference, I have to open the files in Photoshop, zoom in, and toggle which one is visible. If I compute the difference in Photoshop, it's usually only a few levels different, and never more than ~8. That's really hard to see.
Sadly, they're not even using a modern codec like Vorbis or Opus.
That said, while I doubt I can tell the difference between FLAC and high-quality Vorbis, once I learned to listen for distortions on the high end, it's a little hard to unhear. Same as looking for blocking artifacts in JPEGs or video.
FLAC is lossless, and given a random piece of audio it would be impossible to always encode into 700kbps something that is originally 1.4Mbps without loss of information. So your 2:1 samples claim doesn't sound accurate. Maybe file size, but not sample count.
Yes, if you are encoding level-maximized white noise, FLAC cannot compress at that ratio, or likely at all.
Music is not white noise. It does not maximize the channel capacity of the audio spectrum. As a rule of thumb, FLAC compresses most anything a human might call "music" in about a 2:1 file size ratio.
I do experience differences on my HifiBerry + MFB (motional feedback) speakers between 320bps mp3 and flac. Mainly the crispness of e.g. percussion and voices is subjectively much, much better.
This might of course be the IKEA effect, but I'm quite happy nevertheless with my flac collection.
one subtle point they missed: if your headphones are wireless (as it seems many are these days, as are car audio interfaces), there's a fair chance the audio is getting compressed again by a different lossy codec.
All bluetooth is compressed with a variety of algorithms. Some are better than others, but none are lossless (not even aptX-HD). I believe Apple is about to release a firmware update that they claim will allow some of their cans and buds to support ALAC, which is lossless, IF you have Bluetooth 5 devices, and only use it short range. My guess is that it will dynamically degrade down to AAC-256 if the devices are too far apart. Bluetooth 4 devices cannot support lossless simply because they do not have sufficient bandwidth.
I am sitting in a commercial studio today, working on a mix. I have been sitting here for most days over the past 8 years. This is my job.
The equipment in front of me in the control room alone is worth around 90000 (SSL board, RME conversion, Neumann kh410s for monitoring and serious acoustic treatment) — nothing super crazy when it comes to studio setups but certainly in the 'above average' area of things, even in this profession and certainly when it comes to anything normies are gonna experience. My most expensive headphones (I own around 10) are Sennheisers for ~1.5k, powered by a Phonitor, which cost around the same.
To toot my horn, I have been fairly successful at this. I work with chart pop/alt/indie musicians on a regular basis. I am actually in the rare position to have made a good living off of this.
While working on a Mix I am intensely focused on listening, to the point where I will not notice when being talked to. I am listening to and passing judgement on over 100 individual sources of audio running at the same time. I will hear if one of the 16 background vocals tracks is slightly out of tune. I will hear if there's an unpleasant resonance on a guitar somewhere for a split second. Absolutely nobody, including the artist, will ever be as acoustically intimate with the whole thing as I am during this time (this is probably true for any notable mix engineer).
Considering all this, the highly intensive and focused process, the considerable amount of money carefully spent (in contrast to audiophiles just throwing money at something because of what I would imagine are insecurities and boredom, I have to make a profit and thus spent my money carefully even) and my experience, I rank the difference between high quality cbr/vbr mp3 and lossless as so fucking negligible, that I will randomly chose either when making exports to compare my future working status to.
To elaborate on what this means: I will make an export, place it in the project, continue working on it, and then at a later point solo (= switch to) to those exports to check if I am going in the right direction, if I improved on the critical parts, if something got lost on the way.
To be absolutely clear on this: I would cost me an extra half second to make sure I always have a 24/96khz export of that mix as my reference instead of a maybe-i-dont-care mp3. I do not invest that half second. I don't know how to be any clearer than this.
Do not confuse this with me not being able to probably spot the difference. It's just that it is about a skill as spotting a difference between two apparently white walls. I am sure with enough effort and attention to the task anyone would be able to make their brain tune in on the the differences. But why the fuck would anyone do that? There is no upside. No, it will not make you a better music listener or the music more enjoyable. If anything you are doing more focusing on the wrong things.
Every now and then I stumble across this topic online and what used to be insecurity has turned into perplexion. I have no hope for or illusions about that the people who have trained their brains to search for mp3 artefacts (well done, I suppose) but maybe I can reassure some who seem to be on the fence on this topic: Do not dwell. This is a great way to waste time and money and get absolutely nothing of value in return. Aim for headphones you enjoy listening on. They are probably not that expensive. Make sure they have replacement parts available, specially for the ear pieces and cable.
If you are referring to the Tom’s Diner clip, I actually preferred what the 320 mp3 did to the vocal reverb, its a little more stereo/wobbly than in the uncompressed.
How is this still a debate? Just keep a .wav backup and convert it to modern formats as they come. At this point I can only assume people listening to actual mp3's pirated the songs.
A worthless experiment. Audio is only as good as the weakest link. All phones will be insufficient for this test. All computer jacks and all monitor jacks will be insufficient. Most headphones will be insufficient.
Only upon using a dedicated headphone amplifier with a moderately good pair of headphones should the answers become easy.
There are also cases where the differences in amplifiers are obvious - for example some headphone/amplifier combinations have issues with low frequency response (related to impedance IIRC)
Isn't that kind of the point of this being posted? It shows that you don't really need lossless audio on your phone if you need special equipment and an otherwise silent room to tell the difference.
I didn't notice any difference between the three files in any of the cases...and that leads me to question:
Can we verify that the website choices are in fact legit? i.e. the 320k mp3 is in fact 320k and the uncompressed wav is in fact that. And also, that the browser isn't messing with the choices either (by e.g. caching the first file you play and reusing it for the other choices).
Also, I ordered an optical cable so I can use my amplifier's DAC rather than the onboard soundcard DAC.
i'm sitting in front of a 3.5k playback system, decent room, and i produce music for a living and i work in a lossless daw every day, and i did horribly :(
I took the test several months ago with my nice DAC/amp headphones and got 5/6. Just tried again on my phone and got 0/6. And at the end it asks you what you were using.
320 kbps mp3 vs lossless 16bit WAV (cd quality) is extremely difficult to hear the delta without training, but if you know what the source instruments are supposed to sound like, such as hats and you can focus on their frequency band, you can hear compression artifacts and comb filtering.
This course really changed how I take in music and sounds in general.
[0] https://online.berklee.edu/courses/critical-listening-1