MP3 vs. AAC vs. FLAC vs. CD (2008)

https://www.stereophile.com/features/308mp3cd/index.html

by thefilmore • 3 years ago

146 371 3 years ago

torrance• 3 years ago

2008.

Utterly useless. No listening tests except for a custom made sound file which is designed as an artificial worst case for the codecs. You might as well benchmark a text codec on /dev/random...

kevincox• 3 years ago

Yeah, this feels like someone arguing against no-one. They just showed that lossy codecs are indeed lossy. I don't think anyone was arguing that MP3 isn't actually lossy. (They did have a bit of interesting info about where the loss creeped in but that seemed to be a side node to the "argument".)

The question is if humans can hear the difference in the lossy waveform and if that harms the listening experience.

drupe• 3 years ago

No, you can't hear the difference, let's collectively move on, please.

mjlee2• 3 years ago

To be fair, that's true today, but not in 2008 when this article was written. MP3 encoders have come a long way and bitrates are typically much higher.

swagempire5• 3 years ago

I'm not sure its true even today. I've done lots of work in studios and can hear the difference between MP3 vs FLAC and CD -- even these days.

amluto4• 3 years ago

…if competently encoded.

You can absolutely hear the difference between a bad MP3 and the original. I used to amuse myself and friends by quite reliably identifying the difference, blinded, using a rather bad pair of speakers.

Actual CD audio can also work quite differently than any encoding, as at least older CD drives had an entirely separate analog output cable that connected to the sound card and bypassed the ATAPI link entirely. Levels wouldn’t even be matched.

hnlmorg• 3 years ago

I can definitely hear the difference on some songs at some “CD quality” bit rates of MP3. Also some MP3 encoders (and decoders too, to be fair) are better than others. Particularly back when this article was written.

That all said, these days encoders are much better, and there’s no excuse not to go for 320kbps (assuming you have to use MP3).

What I find more interesting is that there was a period where some people who grew up listening to MP3s preferred the artefacting they introduced vs lossless. In much the same way how vinyl enthusiasts like the colouring of the sound that medium introduces. Which just goes to show that as much of this is down to psychology as it is technology.

throwaway1672• 3 years ago

With a headline like that it felt like 2002 again.

The article's byline has 2008.

A 2023 update could be interesting comparing the streaming providers' choices, and persistence of choices, now that monthly subscriptions, rather than actually owning anything, are so dominant.

deciduously• 3 years ago

Citing his own past work from 1995, too.

esafak2• 3 years ago

I'm glad it's an old article. It would be sad if those audiophiles were still debating this old chestnut.

Tsiklon• 3 years ago

I’ve been involved in some heinous forum threads with people arguing the difference in sound of music streamed from different SD Cards. People will argue about all sorts of drivel

Waterluvian• 3 years ago

Maybe someone can make an online quiz of a bunch of formats and accumulate statistics on how well people can actually tell or if it’s just a bunch of random.

jwatzman• 3 years ago

There are a bunch of online tests that help you see if you, with your ears and equipment, can tell the difference. http://abx.digitalfeed.net/ and http://abx.digitalfeed.net/list.lame.html was on HN some time ago, for example. I'm not sure if any of them collect stats though.

conradfr2• 3 years ago

I have an abx website as a side project.

Feel free to make one or take an existing one like https://abx.funkybits.fr/test/the-eagles-hell-freezes-over-h...

mjlee2• 3 years ago

But what about when the codec is dealing not with a simple tone, but with music? One of the signals I put on Test CD 3 (track 25) simulates a musical signal by combining 43 discrete tones with frequencies spaced 500Hz apart.

Yes, but what about with music?

kristofferR2• 3 years ago

Compressed music had a place, but in 2023, with 5G, unlimited data and streaming services, there's really no reason not to go lossless.

ALAC/FLAC files are pretty small, there's few downsides to going lossless. To be fair, there arent that many upsides either, but you at least skip one recompression step when sending the audio over BT.

amluto4• 3 years ago

Have you contemplated what AWS charges to egress a music file in 2023? Gigabytes are expensive.

kristofferR2• 3 years ago

I don't see the relevance to AWS regarding people's music habits.

tjoff2• 3 years ago

The bigger question is why people uses AWS for sending bulk data.

Gigabytes are cheap.

almatabata• 3 years ago

Flac takes a lot of space on my drive. Most people will not want to have that much data on their drive just for music. If i look at my folder now:

Yanni Rainmaker Flac -> 40 MB Yanni Rainmaker Mp3 -> 3MB

More than a factor 10 for a single song. For 50 songs that would become 2Gb. I love flac as a format but i would never recommend it as a general format for my grandmother.

mixmastamyk2• 3 years ago

I was suffering from low disk space for a few years and then happened to notice last month that 2TB NVME SSDs are $75-$125 depending on speed. They are all much faster than an old drive.

If you haven't looked in five years (like myself) I recommend doing that. No one needs to suffer on short disk space any longer. Don't know what "grandma" uses but it is unlikely that audio is a significant burden anymore when people routinely shoot HD+ video.

Also if compressing, Opus sounds better and is smaller.

ndriscoll• 3 years ago

A 1TB microsd card can store 2-3000 CDs worth of FLAC files. Or an 8TB SATA SSD can store 10s of thousands of CDs. You basically can't fill up a modern drive with (legally acquired) music.

eviks8• 3 years ago

The files aren't small, storage size on a phone is still small especially when photo/video is competing and is taking more space, so the same real downside remains, and there are still plenty of places without any fast data connection even when most of the time you have 5g. Re recompress- ok, but is it a real downside, can anyone notice?

jrajav3• 3 years ago

Let's go over this one more time:

- Q: Can you hear the difference between CD-quality lossless audio and anything higher fidelity? A: No, no one even has the biological ability to. 44100khz, 16-bit audio can perfectly reproduce audio as far as we can physically tell. The only reason to store anything higher is for production or archiving (that is, for computers to listen to).

- Q: Can you hear the difference between 320kbps MP3 or the equivalent, and CD-quality lossless? A: Yes, this is _theoretically_ possible. However, many well-controlled listening tests have been performed on this subject that all say no, so it's much more likely that you can't, and the burden is on you to prove otherwise with an abundance of evidence.

e.g.: https://downloads.bbc.co.uk/rd/pubs/whp/whp-pdf-files/WHP384... https://www.researchgate.net/publication/257068576_Subjectiv...

The listening test linked in the article leads nowhere, I would have liked to see their methodology.

pkulak• 3 years ago

I’ve personally ABX tested people who swore up and down that they could spot 128 AAC, even, and they couldn’t. Never found anyone who could. I know they exist, but they are rare, and probably not the folks who say they can.

Espressosaurus• 3 years ago

There's probably an element of the quality of the DACs and speakers you're using too. If it's a subtle difference it's unlikely we're going to notice it being played through some low-end computer speakers.

jerf• 3 years ago

Some of it is people ask the wrong questions. On a loudness-war-wrecked pop song I may not be able to tell 128Kbps from the original, but on specific content I have been able to tell. I'm not even claiming golden ears or anything; some specific audio content is the audio equivalent of visual confetti [1], and anyone can hear the difference, because the codec isn't even close. And let me underline, I mean, anyone. No special claims being made here.

But all in all, that content is relatively rare, and generally transient even in the music they appear in.

[1] https://www.youtube.com/watch?v=r6Rp-uo6HmI

mattgreenrocks• 3 years ago

128k AAC is quite good, and is roughly akin to 160k MP3. Personally, past 160k on MP3, it gets very hard for me to distinguish bitrates, so I ripped at VBR, averaging at around 200k.

128k MP3s, though, fall apart with more complex instrumentation.

Syzygies2• 3 years ago

"You" is excellent wording here. Who's listening?

I've heard the resonable assertion that the most gifted audio engineers in the world cannot distinguish 192 kHz sample rates from a raw line feed, but some can distinguish 96 kHz. I certainly can't. I used to build audio equipment. There were legendary "golden ears" that people would drive hours to meet, for design feedback. Whatever they heard was reproducible blind with other "golden ears".

How does this square with the logical assertion that there's a sharp cutoff to our biological ability to hear frequencies? Those tests don't account for our ability to sense the presence/absence of overtones.

Now, computers may want better resolution, not to "listen" so much as to better transform in novel ways. Just as "DogTV" recolorizes for its audience, a computer could make the inaudible audible in novel ways. Reconstructing better 3D sound stages. Accurately reconstructing a singer's facial expressions via AI video, rather than simply splatting out something plausible. The whole point of computers are to extend our reach, coevolving in every conceivable way.

mrob7• 3 years ago

Those tests don't account for our ability to sense the presence/absence of overtones.

Overtones are frequencies. If you filter the higher frequencies you remove the higher overtones. The only way you're going to "hear" ultrasound is if it's very loud and you hear audio-frequency distortion generated in your own ear, e.g. like when bats are squeaking nearby. But this isn't useful musically, because everybody's ears distort differently.

miav• 3 years ago

Unless I'm reading it wrong, your second source does very much imply some people can tell the difference quite reliably. As expected, regular people can scarcely tell the difference, but musicians are better at it and sound engineers are in fact quite accurate.

This matches my own experience well: most of my friends do not care about various levels of compression, nor what headphones they use - that's fine, I'm glad they're enjoying art in their own way - but I, and some others, do in fact stand to benefit from less compressed audio.

I've personally done blind tests on myself using a python script that randomly plays compressed and uncompressed snippets of the same track and mp3@320 was not transparent to me (though opus@256 was).

Can I tell the difference when casually listening? I don't know, but when the cost of lossless is having my music collection take 60gb instead of 20gb on my 512+gb device, I have no reason not to go for lossless.

jrajav3• 3 years ago

Examine Figure 1 - The key is the 4th and 5th columns there, CD/256 and CD/320. The results show no significant ability to discriminate between them.

high_priest2• 3 years ago

The thing about being or not being able to point out differences in audio quality is that it all boils down to pattern recognition. If you know anything about pattern recognition, you understrnd that you can't have pattern recognition without prior training through provision of tagged samples of such patterns.

If you would give high quality audio experience, to a person that has been listening through 80s general store headphones, to low quality radio rips on magnetic tapes, you might be surprised how few people are going to describe one as "better", without prior description of work and technology required to produce each experience.

And one would be even more surprised by how many people choose the cassette tapes because of nostalgia and a long time satisfying experience.

green7ea• 3 years ago

I've done the same blind test with decent but not amazing headphones (HD590) and I could tell the difference all the time as long as the music was slightly complex.

If loudness was artificially boosted, I had a harder time but could still often tell. I think the sound engineering of the music played a big role and a lot of modern music isn't mixed with complexity in mind.

I also go for lossless ;-).

analog31• 3 years ago

I created some computed waveforms for audio testing on my PC, and on a whim, stored them as both WAV and MP3. Counterintuitively, the MP3's worked just fine for all of my tests. I didn't dig into the reasons why.

kstrauser• 3 years ago

At its core, an MP3 says that for the next slice of time, play these frequencies at these volumes. If your waveforms are simple, an MP3 can encode them perfectly.

bayindirh5• 3 years ago

The thing is, the “heard” difference between 320kbps MP3 and CD quality & CD quality and higher resolution formats are not “details” per se.

The audible difference can be described as sound stage size, instrument separation and atmosphere.

The problem is making these details audible needs a good system. “Good” doesn’t mean $10K+ here. Two high quality, large-ish two way bookshelf speakers and a good amp with enough punch (50W+ Yamaha or similar will do) plus a good source in a sizable room is enough.

There’ll be people who can’t tell any difference, there’ll be people who can “feel” it, and there’ll be people who can pinpoint differences. This is because the ear training and biological limits of said people.

I have a friend who can pinpoint a half note (natural vs. sharp) mistake in a 90+ people symphony from YouTube recordings, incl. the instrument. His natural ear resolution is around 1/9th of the tone. He always tunes his instruments via ear and verifies with a tuner. So, this is not impossible.

My ears are not that absolute, but I can divide music to layers and pinpoint details, for example.

Lastly, taking a “diff” of CD quality and 320kbps MP3 version of the same track will leave an audible residue.

There are other comments I left over the years here. Search them for more info. I’m on mobile. I have no practical way to link all of them.

duped• 3 years ago

This is the kind of snake oil that double blind studies in controlled reference environments have established is snake oil.

Like for example, the fact there is an "audible" diff is meaningless. The threshold of hearing is not linear nor frequency independent. This is called "masking" and it's exploited by lossy codecs to allow for better encoding as well as audio watermarking. You can add a noise that would be perceptible by itself to content that is entirely masked by the content itself. And the reverse is true, you can remove content without it being perceptible.

liquidise• 3 years ago

I have much of what you describe: 3 listening environments between near field computer speaker setup, reasonably high end home stereo and dac/amp with high end iems.

I have tried multiple times to discern flac vs 320 mp3 across genres. Every time I believe I can figure it out and I consistently fail to exceed 50% (pure chance) accuracy.

Makes me wonder what ultra-linear source gear or speakers would highlight the differences in real-world situations, if at all. But for my purposes I’ll happily accept the roughly 80% file size reduction for no audible difference.

crazygringo• 3 years ago

44100khz, 16-bit audio can perfectly reproduce audio as far as we can physically tell.

I agree on the kHz (as well as on MP3), but I deeply disagree on 16 bits.

Because yes, if you keep your headphone volume at a single reference level and never turn it up, then 16 bits is fine. This is very much proven.

BUT this ignores the fact that people often turn up the volume a ton to hear the quiet part of the classical music, or on that YouTube video where the volume is inexplicably 5% as loud as it should be.

So in practice, 24-bit audio allows you to retain perfect fidelity even when you have to turn the volume up. 16-bit doesn't.

I don't understand why nobody ever talks about this. (Or why you have to install special utilities on your Mac to be able to turn up the volume to 200% or 400% in order to listen to those YouTube videos that are maddeningly recorded at 5% volume.)

jrajav3• 3 years ago

You're right, it's true that 24-bit reduces the noise floor and extends the dynamic range available. However, 16-bit audio already has a range of -96db (for reference, a quiet recording studio typically has an ambient noise floor of around -60db). In practice, this is beyond the noise floor of even the very best hi-fi systems. As you turn the volume up, you will start hearing the noise floor of your equipment long before you hear the noise floor of 16-bit audio.

Unless you mean that 24-bit allows for representing audio that is stored at an extremely quiet level at the peaks, wasting most of the dynamic range. That would make more sense - but if audio is printed in such a flawed way, I would expect other quality issues to be present as well.

kevin_thibedeau3• 3 years ago

There's nothing inherently weak about the fidelity of 16-bit audio on its own. PC audio subsystems don't deliver the full dynamic range on a single audio channel by default. They reserve headroom so that they can mix additional audio sources with less risk of clipping. Audio players that let you increase the volume beyond 100% are just letting you use the full range.

None of this is relevant to a real, dedicated music playback system that doesn't contain a digital mixer. You can't hear noise at -96dB. Your amplifier will swamp that with it's own internal noise sources. In the 80s the audiophools loved to complain that CDs were too quiet because their beloved LP noise was supposed to be desirable for some whack reason.

jart• 3 years ago

People talk about dynamic range compression all the time.

tzs• 3 years ago

You aren't considering the listening environment. It is possible for ultrasonic sounds to interact with objects in the environment or with other ultrasonic sounds to produce lower frequency sounds that are in the range of normal human hearing.

There was an even a creepy ad campaign several years ago that took advantage of this. They had a billboard in New York for A&Es new show "Paranormal State" with the tagline "It's not your imagination".

They used an ultrasonic system on the billboard to make audible sounds appear in a small region on the sidewalk but not anywhere else. When people walking along the sidewalk got to that region they would hear a woman whisper "Who's there? Who's there? It's not your imagination".

That system worked by making a single ultrasonic beam that somehow as it dispersed became audible. There are other systems that use multiple ultrasonic beams that produce audible sound via interference where the beams meet.

Many acoustic instruments do produce significant amounts of sound above normal human hearing range. Cymbals for example have nearly 70% of the sound power above 20 kHz. Trumpets with a mute have almost 2% above 20 kHz.

It seems possible then that if you wanted to produce a recording that reproduces the sound you would get if live acoustic instruments were playing in the same environment you might need to include ultrasonics unless you are making a binaural recording.

This does raise the question of what we actually want playback of a recording to achieve. Is a recording of a string quartet when played back in my living room supposed to sound like that string quartet is playing in my living room, or is it supposed to sound like what I'd have heard if I was there when the piece was recorded, or is it supposed to be something else?

(For those who haven't heard of binaural recordings, they are stereo recordings made by placing microphones inside the ears of a model human head so they record the sounds that actually ends up in each ear when something is recorded live for a listener at a specific location in an environment. This page of headphone tests [2] includes a binaural test if you'd like to such a recording).

[2] https://www.audiocheck.net/soundtests_headphones.php

vel0city14• 3 years ago

These ultrasonic playback things are depending on specific interference patterns materializing at certain spots. The design of the signals, shape of the space, arrangement and characteristics of the playback devices all need to be seriously tuned for this.

You're not experiencing these ultrasonics with headphones. You're not going to faithfully recreate some ultrasonic interference pattern in a random room with a pair of tower speakers placed in any arrangement. If anything, you're more likely to just make extra noise that wouldn't have been there originally from those interference patterns happening haphazardly and chaotically, coupled with the fact the vast majority of audio gear isn't designed to reproduce ultrasoincs (they're often specifically made to try and not generate them!)

And even then, by recording in the room with the instrument, you are capturing the resulting ultrasonic interference that you could have experienced if you were there.

soulofmischief7• 3 years ago

Your first point stands up to experimental scrutiny, but your second needs qualification: Anyone can be trained to pick up the differences between 320kbps mp3 and lossless formats.

Compression kills the high end, and learning to recognize tell-tale compression artifacts will forever ruin your ability to appreciate streamed music, low-bandwidth wireless audio systems, or just 320kbps rips of music, certain genres faring worse than others.

c0pium4• 3 years ago

That is absolutely not true. People can be trained to hear that difference when the testing is not blinded, however in triangle tests that ability vanishes.

pja4• 3 years ago

Are we talking mp3 or other codecs here? mp3 has a a couple of encoding ”tells” that a trained listener can pick up on (although it gets increasingly hard to do so beyond 128kbit IIRC). Other codecs don’t even have those & at higher bitrates people can’t pick them out in blind listening tests.

goalieca• 3 years ago

16-bit audio can perfectly reproduce audio as far as we can physically tell.

Imagine encoding a sort of real world dynamic range across 16-bits. This would go from 0db to 100db in volume. This would need more than 16-bits which yields an SNR of about 96db. The dB values are different and not comparable but you can see we don’t capture the full dynamic range of human hearing very well.

mrob7• 3 years ago

Humans don't have 100dB dynamic range across their full hearing spectrum. We're less sensitive to high frequencies, which means you can apply high-frequency dithering to improve the dynamic range without adding audible noise.

https://en.wikipedia.org/wiki/Noise_shaping

ptx3• 3 years ago

What about when the output of the lossy codec is passed through another lossy codec, e.g. MP3 through AAC over Bluetooth? I would expect better results (from the second codec) when starting from a pristine lossless source.

criddell2• 3 years ago

That would be an interesting experiment. Take a hi-res file and encode it with encoder A then B then A then B then ...

How many encodings does it take before a trained listener using good equipment in an ideal setting can tell?

CharlesW3• 3 years ago

What about when the output of the lossy codec is passed through another lossy codec, e.g. MP3 through AAC over Bluetooth? I would expect better results (from the second codec) when starting from a pristine lossless source.

It's true that, technically, you'll get better results from the second codec when starting from the uncompressed source. Generally, it's always better to avoid unnecessary generation loss. That doesn't necessarily mean that you'll hear a difference since that depends on the cumulative output quality.

rplst812• 3 years ago

This is a great example of why I prefer lossless source material. I can tell an MP3 from a CD nearly every time (on recordings of good original quality) once it's sent through Bluetooth.

dTal• 3 years ago

Or just poor listening conditions. Everyone always talks about needing super high end setups to hear the difference, but it's actually the low end setups with wonky frequency responses that totally mess up the assumptions behind psychoacoustic masking.

replete2• 3 years ago

We AB tested 16-44.1 and 24-96 versions of some really good classical recordings recently - you need good listening equipment (ears and electronic) but undoubtedly the dynamic range and top end (particularly) sounded better. It really depends on the listener, the source, and the equipment.

A few years ago I did lots of AB testing with some Sony xm1000w3s (Sony LDAC) and Tidal Hifi with some 24bit masters and it was an incredible experience that changed my mind in the whole "640K.. 16bit is enough" argument.

nyolfen2• 3 years ago

"you need more than anecdotal evidence"

"have some anecdotal evidence"

fsckboy• 3 years ago

things that are slightly louder "sound better". How did you control this sort of thing?

Mistletoe• 3 years ago

Was your test blinded? I guess there is a chance you are an outlier but blind tests like this one don't support what you are saying.

http://archimago.blogspot.com/2014/06/24-bit-vs-16-bit-audio...

In a naturalistic survey of 140 respondents using high quality musical samples sourced from high-resolution 24/96 digital audio collected over 2 months, there was no evidence that 24-bit audio could be appreciably differentiated from the same music dithered down to 16-bits using a basic algorithm (Adobe Audition 3, flat triangular dither, 0.5 bits).
Furthermore, analysis of those utilizing more expensive audio systems ($6,000+) did not show any evidence of the respondents being able to identify the 24-bit audio. Those using headphones likewise did not show any stronger preference for the higher bit-depth sample. No difference was noted in the "older" (51+ years) age group data (not surprising if there is no discernible difference even with potential age-related hearing acuity changes).

eviks8• 3 years ago

I'd rather trust solid hearing biology/physics plus all the other failed tests

the effective dynamic range of 16 bit audio reaches 120dB in practice [13], more than fifteen times deeper than the 96dB claim.
120dB is greater than the difference between a deserted 'soundproof' room and a sound loud enough to cause hearing damage in seconds.
16 bits is enough to store all we can hear, and will be enough forever.

https://people.xiph.org/~xiphmont/demo/neil-young.html#toc_1...

jbverschoor5• 3 years ago

These days "good equipment" unfortunately means:

- Sonos

- Airpods

- Beats

ReactiveJelly4• 3 years ago

Why AB and not ABX?

amlib2• 3 years ago

The 24-96 is different master, some sound engineer just had a field day in the studio and produced a better mix. Repeat the test with a 16-44.1 version downsampled (use something like sox with the ultra high quality resmapler) from the 24-96 version and I guarantee you will not be able to spot any difference compared to the "true" 24-96 version.

deaddodo2• 3 years ago

I love how you can pull out 100 studies and side by side comparisons of recording tools/listening devices much more precise than the human ear that all show this as being flim-flam; and still "audiophiles" will convince themselves to spend 5-25k on specialty equipment that has no effect on their experience.

You're better off spending your money on a bog standard DAC/AMP (feel free to opt for tube even, if you insist) combo running through a pair of decent headphones off of 320kbps MP3/AAC (or FLAC, if you insist) source. Even, if we took your subjective insistance that this specialty equipment improved your experience by .00001%, it's probably not worth the 500-1500% increase in expense.

As to your specific example, I can guarantee you that your Bluetooth codec (LDAC or not) introduced far more sound artifacts than the difference between 16 and 24-bit sound.

eredengrin• 3 years ago

How do you know that the 24/96 and 16/44 came from the same masters? If this isn't controlled for then of course the result might be different.[0]

Also, what is xm1000w3s? I can't find any record of this so I'm guessing maybe it is referring to the WH1000XM3 headphones? Given ldac is also mentioned this seems a reasonable guess as it's a bluetooth model. If that's the case I wouldn't call it "good listening equipment", the default frequency response curve of the wh1000xm3 is incredibly bad, it's barely worth listening to classical music on without using AutoEq[1] or something equivalent (I have a pair and it's much worse than my old Ath M50s which were like half the price). The bass heavy curve of the headphones is far more noticeable than any difference between 16/24 bit audio would ever make.

[0] https://people.xiph.org/~xiphmont/demo/neil-young.html#toc_d...

[1] https://autoeq.app/

asveikau• 3 years ago

No, no one even has the biological ability to. 44100khz, 16-bit audio can perfectly reproduce audio as far as we can physically tell. The only reason to store anything higher is for production or archiving (that is, for computers to listen to).

I'm not an expert, but one claim I saw somewhere is that a higher bit width and sample rate is good for people who are mixing and doing audio processing, even where the final result might get downsampled to 44100 hz and 16 bits per sample at the last stage.

SSLy2• 3 years ago

that's what 'production' in the quoted passage means.

ok_computer2• 3 years ago

24 fixed bit and 32 variable/ floating bit rate masters have more head room that may avoid clipping but doesn’t guarantee that. 48 or 96 kHz is useful for time stretching and maintaining fidelity (maybe other post processing without aliasing).

That is all intermediate formats and doesn’t really say anything about what is best for consumers like the standard mastered cd quality at 16 bit 44.1 khz.

Bandcamp is a cool market because I can download wavs from albums to store on my phone. You can see what people use as masters and its all over the place. There are many 96khz masters around and 24 bit depth is popular.

I have a usb audio IO that supports 192KHz across 8xin+out. Those file’s just clog up hard drives so I figure 96 is good enough for bat music.

Blackthorn2• 3 years ago

Yes, that's for antialiasing headroom purposes during the production process.

Slow_Hand2• 3 years ago

There is one situation where 44k/24 bit and 88k/24 bit CAN sound appreciably different, and that's when aliasing is introduced into the recording, mixing, or the sample rate conversion.

If proper precautions are not taken during the recording/mixing/mastering phases aliasing artifacts can be heard in the recording. This may account for the differences that some people hear when judging whether there are differences between the two. Higher sample rate files are more permissive of aliasing and exhibit less perceptible artifacts. So you're less likely to hear it at a higher sample rate.

The artifacts of aliasing manifest as inharmonic distortion that starts at the top octaves and then folds back into lower frequencies as the effect is intensified. This can be easily perceived by most listeners if it is pointed out to them. It is not a pleasant effect like first-order or second-order distortion. It does not compliment the record at all.

That said, if proper precautions are taken to mitigate latency artifacts during the record-making process then a listener shouldn't perceive any difference between a 44k and an 88k record. The best case scenario is often a record that's recorded, mixed, and mastered, at high sample rates, even if it's ultimately be down-sampled to CD quality (44 kHz).

ReactiveJelly4• 3 years ago

So if you had an 88k recording, you could run it through a well-known anti-aliasing filter to create a 44k recording that sounded the same?

So the only situation where 44k and 88k can sound wrong is if... the 44k file is different and wrong?

Slow_Hand2• 3 years ago

Correction: In the last paragraph of my comment I mistakenly typed "latency artifacts" when I should have said "aliasing artifacts".

toast02• 3 years ago

Aliasing is also icky if you go from 44.1kHz to 8kHz (for phone systems), a 48kHz source would be better for that.

But 44.1kHz worked better in the lead up to CDs (works for modulating onto video tape in PAL and monochrome NTSC), so it won until DVD audio brought 48kHz to the masses.

lizknope• 3 years ago

About 20 years ago I bought some live concert DVDs but the band did not release an audio CD version of it. I wanted to listen to it in my car so I ripped the audio from the DVD but when I made a CD it sounded a little off.

Then I learned that DVD was 48 kHz and not 44.1 kHz so the conversion program I used didn't account for this. I went back and used a polyphase filter to adjust the sampling rate. It sounded normal again but there were some audio glitches at various times.

I went back about 10 years ago and ripped it again and converted it to FLAC which supports multiple sampling rates like 44.1, 48, 96, whatever and now everything sounds good.

beebeepka3• 3 years ago

"The human eyes can only do 24 fps" and "if I can't do thing, then no one can" all over again.

zepolen• 3 years ago

No one ever said the human eyes can only do 24fps.

daneel_w6• 3 years ago

You can absolutely hear the difference between 44.1 and 96 kHz sample rate. Even with typical reproduction filters on the output, sampling at 44.1 kHz prevents you from accurately preserving e.g. a sine tone at above ~6 kHz, and that's even without taking into account all the aliasing problems you're facing when the samples don't align with with the peaks of this or that tone. 44.1 kHz is "good enough", but it's not accurate, and you can definitely tell the difference.

As for anything beyond 16 bits amplitude on line level, no, you cannot hear a difference. For such a low-voltage signal the resolution at 16 bits is so fine that it already drowns in all the natural noise and THD in the cables, in the amplifier, in your speakers/headphones etc.

pja4• 3 years ago

Please do watch & internally digest the explanatory videos at https://xiph.org/video/

It explains why you’re wrong in easily digestible terms & how a 44kHz sample rate will accurately encode signals right up to the Nyquist limit. The second video is an end to end demo showing the process in action.

temp0826• 3 years ago

It's been a long time since my DSP classes at uni, but I don't think this is true. 44.1kHz sampling is enough to reproduce up to 22.05kHz sound accurately without aliasing. Unless there is another type of distortion you might be picking up. This stuff is pretty far out of my realm these days.

https://en.wikipedia.org/wiki/Nyquist_frequency

vel0city14• 3 years ago

sampling at 44.1 kHz prevents you from accurately preserving e.g. a sine tone at above ~6 kHz

This is mathematically false. A 6kHz or 8kHz or 10kHz or 20kHz signal absolutely can be perfectly preserved with a 44.1kHz sample rate. Not just kind of preserved, but perfectly preserved.

ReactiveJelly4• 3 years ago

Yeah this is the "stairstep vs. lollipop" thing again.

eyegor• 3 years ago

You should watch this: https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB to better understand why high sample rates are a waste of time, instead of just reading about nyquist. "accurately preserving a 6kHz sine wave" sounds a lot like you think that sample points are reproduced 1-1 from the digital to the analog domain.

This just builds on the xiph video someone else linked but essentially

- sine waves are fine as long as you have points for rising and falling edge (nyquist, 44k guarantees 22k sine wave reproduction)

- bit depth only really affects noise floor, so it depends on your audios dynamic range

sircastor• 3 years ago

And it’s worth noting that if you’re of a certain age - and generally that age correlates closely with the ability to afford equipment that can reproduce the very high quality you’re demanding - your hearing has likely deteriorated past the ability to discern the difference.

mrob7• 3 years ago

You don't need super high-end equipment to hear subtle details in audio, you just need reasonably good headphones. A few hundred dollars worth of headphones will get you quality that would cost tens of thousands with speakers and the room treatment speakers need to perform at their best (digital room correction can make a good room sound great but it can't fix a bad one).

saaaaaam4• 3 years ago

Some time ago - though not as far back as this article was published - we did an experiment at a conference that we held in the demo facilities of a Very Well Known Audio Company.

We played a range of snippets of music - rock, classical, electronic, pop - at various qualities over what was quite possibly the best sound system in the world.

The audience was a significant number of record label executives, distribution execs and general audio/music industry experts.

We played pairs of the same snippet and asked people to tell us which was higher or lower quality.

One person got them all correct. Turned out he’d mastered one of the early tracks we played so had a good reference and then used that as a baseline for the others.

Everyone else it was completely scattershot.

It wasn’t a controlled experiment but it was definitely interesting.

lmm6• 3 years ago

This is why an ABX test is the way to go. If you don't know which version of a snippet is "right" there's no way to objectively say which one is "better" - maybe you like the distortions (cf the famous vinyl "warm sound").

zigzag3125• 3 years ago

asked people to tell us which was higher or lower quality

This test didn't measure what you probably wanted it to measure.

justsomehnguy3• 3 years ago

Some time ago - just a couple of years ago, I decided to get back to the roots and to listen some Iron Maiden. After a visit to the Brown Sector I made out with the full discography in FLAC but also I got a copy of Powerslave in AAC ripped from iTunes.

And all I can say what I really, really hear the difference.

Because iTunes version is mastered to sound good in iPods so it has a quite noticeable bass boost all over the album, which is extremely noticeable on my 2.0 acoustic which itself has a good bass boost, so this version sounds quite muffled compared to the FLAC version from the CD (CP32-5043).

But on the go, with my CX300-II / CX3.00 there is no noticeable difference.

keeganpoppen2• 3 years ago

wait, but if he could distinguish them from having mastered one of the tracks, doesn't that kind cast doubt on the "no human could possibly ever distinguish these things" mantra? as an aside, am i alone in feeling like the absolute certainty and hyperbole with which this argument tends to be stated potentially has a bit of a "doth protest too much" vibe to it? i definitely know it is a minority view, and that is ok-- i'm happy to be wrong and skeptical.

skeptical though i may be, i'm definitely not here to say that "audiophiles" aren't charlatans or anything like that, for the record. and while i don't totally understand the setup you describe in the sense that i don't get why insider knowledge on one track would tip all the rest of them (were the HQ tracks played either all first or all second, or something?), wouldn't one person's ability to completely discriminate between the two encodings seem to be very strong evidence that it is possible to tell the encodings apart? the kinds of differences between master recording and 16bit 44.1 kHz are exactly the kinds of things that would give away which encoding is higher quality, no?

i feel like there is this moving target thing that goes on sometimes, where the strong argument made loudly is "no human can tell the difference", and then the tests are more like "most people can't tell the difference between things that they have no reason to be attuned to well enough to have any chance of picking up these subtle differences.

forgive me if i misrepresent, or come across like some sort of audio quality chauvinist-- i ask all of this in earnest, and without having a strong opinion one way or the other.

nix0n• 3 years ago

The listening test linked in the article leads nowhere, I would have liked to see their methodology.

Here you go:

https://web.archive.org/web/20080322114622/https://www.stere...

fuzzfactor2• 3 years ago

Here's the full results from 2013 by the original blogger:

https://archimago.blogspot.com/search?updated-max=2013-02-24...

Starts about halfway down the webpage.

ScoobleDoodle• 3 years ago

What quality and power speakers are needed to get good output from the files so it can be heard?

m463• 3 years ago

I wonder how far we still have to go.

Computer graphics is pretty good, but how does it compare to walking out into a bright sunny day.

Audiowise, I wonder how listening to live music, then listening to something that went through capture and playback end-to-end.

I'll bet there are differences and I wonder where the "bottlenecks" are.

vel0city14• 3 years ago

If you're listening to most professional live sets that has amplifiers you're listening to something that went through "capture and playback end to end". Loads of boards are digital these days, and you're probably going to be altering the mix and EQ and going through compressors and all kinds of stuff in between.

dghughes• 3 years ago

Can you hear the difference

Do you think it matters if I play the song on my $10 cheapo earbuds or on $60,000 Sennheiser HE-1 Summit headphones?

brewdad2• 3 years ago

On $10 cheap buds? Yes. On $100 middling buds? Very, very few people will notice a difference.

joshspankit• 3 years ago

I wish years ago we would have switched to A/B testing taking weeks or months for each side.

dwroberts• 3 years ago

Worth noting 48KHz audio is now a commonly encountered standard for video. Not to say it’s necessarily audibly discernible from 44K but it’s obviously not quite as straightforward as 44K being the end of the story.

lmm6• 3 years ago

48 is just about convenience, it's not meant to be "better" than 44. 45, 46, 47 or 49 would be fine too, but 48 is a rounder number.

mrob7• 3 years ago

48kHz has the advantage of being an integer multiple of many common video frame rates, which makes video editing simpler.

danbee4• 3 years ago

48Khz can theoretically be better because the anti-aliasing and reconstruction filters can have a gentler slope, but modern oversampling converters make this moot now.

rplst812• 3 years ago

Agreed on the 44.1k 16-bit is all you need as a delivery medium for a final audio product.

However higher bit rates and sample rates are needed for multi track recordings so that during the mixing stage and mastering, the fidelity is preserved when _math_ causes rounding errors and what not. Unless, you are using nondestructive editing.

As for listening to the final product, i.e. store bought CDs and their equivalent MP3 and AAC rips... I can often hear the difference in specific recordings, no matter the bit rate, because the perceptual encoding schemes often butcher certain recordings.

For example, on RUSH's Red Barchetta from Moving Pictures, there is a synth intro that slowly vamps in volume. Every MP3 encoder (that is normally worth it's salt) I've ever tried encoding that with outputs a garbled, distorted, electronic sounding distortion. It clears up immediately once it reaches full volume, but during the crescendo it falls on it's face.

jtriangle• 3 years ago

If you know what to listen for in the high frequencies, you can pick out mp3's pretty easily, provided said mp3's have that sort of content.

The question really isn't 'can you tell', it's 'does it matter', and, well, most of the time, no, it does not, even for lower bitrate mp3's.

There are many people, of course, that don't like the idea of lower quality audio, and they can tell at least sometimes, so they 'dislike' mp3 in general. That's all well and good until they start saying silly things like 'mp3's sound bad', which is not true in any sense.

MuffinFlavored• 3 years ago

Can you hear the difference between 320kbps MP3 or the equivalent, and CD-quality lossless?

I find it hard to believe an entire industry exists with high-end audio equipment ($100k+ on speakers/receivers/room treatment) just to play 320kbps MP3s?

vel0city14• 3 years ago

The poster was talking about digital formats and compression. The expense of speakers and receivers is usually higher quality analog reproduction. Two vastly different concepts at play there. Even then going into the $100k+ territory usually yields very little results on top of the many thousands of dollars setup. Its probably better spending at least half of that just making the room and the other sound reflecting/absorbing things better if you're going that high-end.

RockRobotRock2• 3 years ago

It's also hard to believe homeopathic medicine exists.

ekianjo• 3 years ago

Can you hear the difference between 320kbps MP3 or the equivalent

While I agree with the premise this also depends massively on the equipment used to reproduce the sound. If you have a good amp and large speakers you are more likely to notice than your cheap headphones or thru your crappy laptop speakers.

dclowd9901• 3 years ago

I see audio storage like I see cars: there’s no such thing as a “best” version — more like the right version for the right time.

On a plane, I’m glad I can pull down a ton of albums on Spotify and listen to them without access to internet connectivity.

For Eno, Steely Dan, Roxy Music and Thomas Dolby produced albums, I’m glad for lossless.

For sitting at home on a Friday evening with a bourbon in my hand and the lights down low, nothing beats my vinyl player.

I could get into the cars side of it too but that’s more a “me” thing than a “relevant to this conversation” thing.

zigzag3125• 3 years ago

The MP3 study you referenced uses flawed methodology:

participants were asked to listen to both versions as many times as needed and to choose the version they preferred in a double blind A/B comparison task

I have trouble finding high quality studies comparing 16-bit to 24-bit audio. This one one is kind of interesting:

https://www.researchgate.net/publication/338989993_Study_on_...

gabereiser• 3 years ago

We all know the Vorbis is supreme. Get out of here with your 15 year old DRM compression riddled subpar listening formats. OGG is all that matters. Without it… we wouldn’t have Spotify. <leaves before shoe is thrown>.

mikeytown2• 3 years ago

Didn't use lame for mp3 so the conclusions are pointless in my opinion

thriftwy• 3 years ago

Lame is the most popular encoder so it is highly likely you have listened to its output.

ck45• 3 years ago

Let's assume there are people who are able to hear a difference. Why does it matter to a majority of people and the way they consume music? Maybe I'm from a spoiled generation, growing up listening to FM radio and tapes, even copying from tape to tape.

A lot of rock music lives from the imperfection of audio equipment, people spend a considerable amount of time replicating the behavior of vacuum tubes. Even techo producers like Robert Babicz record to analogue tape machine to enhance the final result.

blipvert• 3 years ago

Quite literally the sound of rock music is the sound of distortion. The kinks didn’t razor blade their amps for no reason.

lifthrasiir3• 3 years ago

We recommend that, for serious listening, our readers use uncompressed audio file formats, such as WAV or AIF—or, if file size is an issue because of limited hard-drive space, use a lossless format such as FLAC or ALC.

I recommend that, for serious listening (for some weird definition of "serious"), go to a music concert. PCM is also a lossy compression due to the quantization step, albeit its effect is much less pronounced for so many reasons that no one even thinks it as a "compression" method. If you can tolerate PCM, you should be also able to accept some good enough lossy codecs---I don't know if that includes MP3 or AAC or Vorbis or Opus or whatever, though.

jraph• 3 years ago

I think it depends on the style of music. I guess for orchestral / classical music it's good.

But for other styles, I don't enjoy concerts for audio quality.

It's usually way too loud, so you have to wear earplugs. I've heard some made for this don't skew audio too much, but they are still a filter.

And then you have to like the balance that's chosen by the audio engineers and they are often not ideal. The voices can sometimes be not loud enough to the point you don't hear the words well, the bass too loud. Frequencies don't all travel the same way, so if you are too far away some things are missing or distorted, etc.

And then there's the noises from other people, the claps, the screams, etc.

And the audio still possibly went through some kind of non-analog equipment.

Not saying that feeling the bass in your whole body and feeling the communicative / excited atmosphere from the crowd can't be enjoyable but for audio quality, I'd rather listen to music in a calm room with some good equipment, at a volume level comfortable to me, when audio engineering didn't have to be live and could be (even) more carefully managed.

If you can tolerate PCM

Are there people who can't tolerate it? It must not be very convenient.

(Huge caveat to this comment: I listen to music most of my awaken hours, but I'm not an audiophile. I never carefully listen to music, it's usually in the background.)

akira2501• 3 years ago

PCM is also a lossy compression due to the quantization step, albeit its effect is much less pronounced for so many reasons that no one even thinks it as a "compression" method.

    20 * log10(1.0 / 2**16) == -96db

Much like sampling rate, it produces a range that's most likely outside of the ability for any human to appreciably detect. It's also a constant effect, whereas codecs actually analyze the audio to determine which components of the frequency spectrum it can eliminate.

I don't think it's reasonable to compare PCM and lossy codecs this way.

rplst812• 3 years ago

PCM is also a lossy compression due to the quantization step

Wrong, especially today. Modern ADCs use oversampling to push quantization noise into the inaudible range, and then filter it out before decimation to standard PCM.

adrian_b• 3 years ago

Sorry, it is what you say that is wrong.

Because the end result is standard PCM, the quantization can be only worse, not better.

Oversampling ADCs push a much greater quantization noise into the inaudible range, and then, by low-pass filtering, reduce the quantization noise to the level of standard PCM.

Oversampling ADCs are not better, they are much cheaper, because 16-bit or 24-bit quantizers with enough speed and accuracy are extremely expensive.

Oversampling, i.e. sigma-delta modulation, in both ADCs and DACs, allows the use of much cheaper quantizers with low resolution, of only a few bits, or even of only 1 bit, and of much cheaper filters, which do not have to be very steep, without degrading too much the quality in comparison with real PCM conversion done at the Nyquist frequency.

deadbeeves5• 3 years ago

PCM is also a lossy compression due to the quantization step

No, it's a lossy encoding step. Losing at least some of the information of a performance when you record it is unavoidable. For PCM to be a lossless, when you play it back it would have to transport you back in time to when the performance was recorded and you should be able to touch the performer. You're being silly.

lifthrasiir3• 3 years ago

That's a silly answer for a silly claim, if I really need a clarification.

marcodiego• 3 years ago

Beyond comparing formats, even subjectively, it is important to consider how the public got used to compression artifacts.

I mean, the famous mp3 pre-echo was so common in early 90's that I think part of the listeners would prefer listening to it than to a cleaner sound. It is possible that mp3 influenced how music is composed, mastered and mixed.

That being said and adding the fact that people are willing to listen to music using cheap auricular phones in the noisy environment of their cars and recompressed using Bluetooth, I'd say that the 128kbps mp3 is still a very hard to beat format.

Isthatablackgsd• 3 years ago

Not surprising about the MP3 pre-echo. I remember reading a comment while back, I think it was in Reddit. Someone posted a comment about how they bought a CD that have 50s/60s music for their grandparent and gave it them as Christmas gift. Their grandparent are graceful about the gift but they didn't use it very often. They inquired about why they are using the vinyl over a crystal clear CD. The grandparents said they loves the hissing and other sounds that are prevalent on the vinyl. They said it made it part of the music and they have strong memory of it during their era. The CD completely removed those sounds and felt it is unnatural sounds they are not used to. They felt it is too clean.

People have strong relations with their musics that they are used to while growing up. Nostalgia are powerful memories and they don't want their music unsullied from something that they grew up with.

thefilmore3• 3 years ago

MP3 smearing can be noticed even at 256kbps [1].

[1] https://www.soundonsound.com/techniques/what-data-compressio...

iggldiggl3• 3 years ago

Well, MP3 does have some fundamental limitations that were only fixed with subsequent codec generations, like AAC or Opus.

tamimio4• 3 years ago

I’m not entirely into the audiophile stuff, but from personal experience, you can tell the difference, last thing I tried was when I switched from Spotify to Apple Music, where the later has “lossless” option (I think even Spotify has that) but the difference was clear between the two for a streaming service, Apple one is just more clear and alive, I even opened the same song and kept switching back and forth between the apps just to make sure I’m not imagining stuff. Was it because the lossless on apple is better than the lossless on Spotify? Or something else? I don’t know.

masklinn• 3 years ago

Even though they’ve been talking about it for 2 years AFAIK Spotify still does not have a lossless offer. And lossy bitrate is adaptive and middling (especially if not using premium).

EQ-ing and mastering could also be different.

tamimio4• 3 years ago

EQ-ing and mastering could also be different.

Possible, although I didn’t change any default ones.

teolandon• 3 years ago

There is no lossless on Spotify.

tamimio4• 3 years ago

Well, than that’s definitely the reason why

Scoring6931• 3 years ago

It could be simply a difference in sound volume. Our brains tend to believe louder is better.

tamimio4• 3 years ago

In my humble unscientific test, I did have both at the same volume level (didn’t check the EQ though as what other commenter said), I even tried different speakers incl my car’s, nothing fancy or extremely measured, just as an average consumer perspective.

tjoff2• 3 years ago

Doubt it. The problem is that you don't know the source of the audio. That is the key difference, and since streaming even if you pick the right album the "same" song might come from another because it "is the same".

If you see such a huge difference across all music the playback software have manipulated the audio.

Assuming you have high quality set on spotify (even in the mobile-streaming setting, if you didn't use wifi).

oriolid2• 3 years ago

I have the same experience with Spotify vs Deezer. I think it's more likely that Spotify has somehow screwed up their encoding process than that I would hear the difference between high bitrate lossy and lossless compression. Spotify's volume normalization also somehow makes everything sound worse but it's easy to disable.

raphlinus4• 3 years ago

Why is the standard considered to be CD quality? In that way, the article shows its age. Today you wouldn't be talking about 44.1kHz 16 bit, it would be all about 24 bit 192kHz. If you're looking at spectrum plots, CD is very much on the low quality side of the spectrum of what's possible. Maybe we should be considering megahertz sampling rates and 32 bits, surely we have enough bandwidth.

Why not then? Because there is a ton of science and empirical evidence that humans cannot hear the difference[1]. Good engineering is about meeting the requirements with minimal cost. If the requirement is that it sounds good to humans, and the cost is number of bits to encode (and thus store and transmit) the signal, then modern codecs like Opus are clearly superior to uncompressed and losslessly compressed signals, much less higher sampling rates.

If your goal is something other than good engineering, for example the aesthetic satisfaction that the bits are the same as what the mastering engineer put on the CD, or for some reason caring how clean spectrum plots of artificial signals look, then the arguments may have some merit. But let's be clear on the goals.

[1] https://people.xiph.org/~xiphmont/demo/neil-young.html

galleywest200• 3 years ago

The point is to have all of the information stored in your archive.

You can compress it for listening later, but you can never add information _back into_ the file. Store it in FLAC for archival purposes.

An equivalent would be archiving works of visual art in JPEG and not something lossless.

raphlinus4• 3 years ago

Agreed, for archival purposes we should be using lossless codecs. Not because you can hear the difference but because it makes it easier to reason about whether there's any distortion introduced by the compression. And we can consider the original artifact, as created by the mastering engineer, an authoritative source of truth, even if it's an imperfect representation of what was performed by the musicians.

bobsmooth• 3 years ago

Why is the standard considered to be CD quality?

Because it can fully reproduce everything the human ear can hear. Higher bitrates are only useful for production or archival.

20after4• 3 years ago

everyone should really watch https://xiph.org/video/vid1.shtml and https://xiph.org/video/vid2.shtml

ptx3• 3 years ago

Their recommendation doesn't make any sense. They first explain that lossless compression reproduces exactly the same data as when uncompressed:

"Lossless compression is benign in its effect on the music. It is akin to LHA or WinZip computer data crunchers in packing the data more efficiently on the disk, but the data you read out are the same as went in."

...but then recommend uncompressed over lossless compression for "serious listening":

"We recommend that, for serious listening, our readers use uncompressed audio file formats, such as WAV or AIF—or, if file size is an issue because of limited hard-drive space, use a lossless format such as FLAC or ALC."

croes• 3 years ago

Maybe decoding speed is an issue or decoder quality

ptx3• 3 years ago

The decoder either reproduces exactly the same bytes as the original or it's seriously broken. You can run "flac --verify file.wav" to test this, or compare the decoded bytes yourself if you don't trust the tool. I doubt such bugs are a common issue.

I suppose decoding speed could matter in some situations, but they said "for serious listening", not "if your system is so slow that it fails to decode the file in real time".

eric__cartman• 3 years ago

Yeah, that doesn't make any sense at all. The only reason I can think of to use WAV instead of a losslessly compressed FLAC is if the player being used has a dog slow CPU or an incredibly old software stack that can't play FLAC files.

But I doubt these guys are using a Pentium 1 machine to play their audio files so idk. The low end smartphone I had in 2013 could easily play FLAC files, at least in the real time uncompressing and decoding part of the equation. Now if the built in DAC and amplifier could take advantage of that extra data is another thing.

hollandheese2• 3 years ago

Funny thing is that even on a Pentium 1, it doesn't make much difference. You'd have to go to a 486 to really choose WAV. Even then FLAC is absolutely playable, it just takes a bit more than twice the CPU of WAV (70% vs 30%). MP3 though on the 486 is just too heavy to do.

mrob7• 3 years ago

There are two reasons to use WAV over FLAC in music production: you can load WAV files into your DAW faster; and WAV supports floating point, which means you never have to think about clipping until the final mastering step. Neither is relevant to listening.

danbee4• 3 years ago

FLAC has a massive advantage over WAV/AIF in that it can be tagged with metadata for the music player software to read and display.

wesapien• 3 years ago

Why do you guys tell others what they can or can't hear? We need to have someone do head surgery on you and implant a device to detect signals between your eardrum and the brain to see what's heard. Even then the brain probably has a lot to do with how sound is interpreted or processed.

tommek4077• 3 years ago

You can easily create blind tests and they speak a very clear outcome.

swagempire5• 3 years ago

Anything is better than YouTube-- which seems to be the common format everyone is listening to these days. I would LOVE to be able to regularly listen to CD quality music...

derkades• 3 years ago

YouTube is quite good with >128kbps opus audio

swagempire5• 3 years ago

It's acceptable for most people listening to it on tiny speakers on phones or even earphones. But there is no way it's high-fi.

flashback21992• 3 years ago

For mp3, for me, 192kbps and higher is where it sounds pretty good, 128kbps sounds bad

mrob7• 3 years ago

Required bitrate depends on the music. With a modern version of LAME, 128kbps will be very difficult to ABX for solo vocals, but much easier in busy rock music (specifically, by listening to the decay of the cymbals).

This is why variable bit rate was developed.

flashback21992• 3 years ago

oh no here come the audiophiles runs away

iggldiggl3• 3 years ago

This is why variable bit rate was developed.

A more modern codecs like AAC or Opus, that can e.g. better deal with the cymbals problem you mention.

isykt• 3 years ago

What's really infuriating to me is that, at one point, Apple took a reasonable stand with their music streaming and said "256kbps AAC is CD quality" (it is). And now they've turned around and starting pushing this snake-oil, DRM'd "Hi-Res Lossless" nonsense.

The only reason to have "Hi-Res Lossless" is if you're going to do something besides listening with it... and you can't with Apple's streaming.

RIMR• 3 years ago

OP forgot to put (2008) at the end of this.

Today's standard isn't "CD Quality" anymore. There is literally no audible difference between MP3 320kbps, which covers the complete range of human hearing up to 22kHz and FLAC which covers all the way to 192kHz, which is lossless. At this point digital audio has surpassed what the human ear is capable of hearing, and any advancements to this is superfluous as far as music is concerned.

The only advantage to raw or lossless formats for music is archiving, as FLAC can be converted into other formats without incurring additional quality loss. For listening, it is now more important to have good equipment rather than a lossless format, and for streaming it is generally preferable to keep bandwidth requirements down.

The only reason I can imagine to continue expanding the capabilities of lossless audio is for scientific purposes and machine learning where the limits of human sensory perception isn't a limiting factor.

snvzz2• 3 years ago

The only reason I can imagine to continue expanding the capabilities of lossless audio is for scientific purposes and machine learning...

Communications is the main reason. There's only so much bandwidth in 44.1KHz.

oriolid2• 3 years ago

One of the benefits of lossless compression is that you don't have to put any effort into thinking about what level of compression is good enough and who can hear the difference. It just doesn't make any difference.

iggldiggl3• 3 years ago

There is literally no audible difference between MP3 320kbps, which covers the complete range of human hearing up to 22kHz

Although MP3 does have some fundamental limitations that cannot be fixed no matter how much bitrate you throw at them (referring especially to the "Inoptimal window sizes" from https://web.archive.org/web/20120222124415/http://www.mp3-te...). They're not dramatic issues, to be fair, but as long as you go for lossly compression, using a somewhat more modern codec like AAC or Opus would be preferable, unless you absolutely need the maximum compatibility afforded by MP3 (though these days at least AAC support should be pretty widespread, too, plus the patents on regular ["low complexity"] AAC have expired as well).

ksec• 3 years ago

1. This is an 2008 article. Per Guidelines you should put years in the Title.

2. MP3 has improved a lot over its lifetime. LAME was already used for default by year 2000. When people say MP3 was good enough, they refer to MP3 encoded with LAME. ( Rant: When we people learn the codec, encoder and the encoded results are different things? 2023 and I see this mistakes everywhere still )

3. Even iTunes AAC has seen lots improvement since 2008. Especially in the 256Kbps+ Range.

4. And when AAC is mentioned. That is AAC-LC ( Or AAC Main Profile which isn't all that different ). AAC-LC ( Low Complexity ) has been declared as Patent free by RedHat. There is no reason to use MP3 today.

5. The definition of "CD-quality" alike went from MP3 128Kbps to now AAC 256Kbps. And arguably that is true for consumer market. Even Hydrogen audio has repeated these test multiple times.

6. I still prefer the codec MPC, Musepack (https://www.musepack.net). Sorry I just had to write it out. Sadly it never gained any traction.

7. If we have to be picky about frequency range, may be CD itself isn't good enough and we could use SACD?

8. Lossless is making a come back. Storage and Bandwidth cost continues to fall. ( Arguably not true for NAND, but let's ignore that part for now )

9. It is ironic when Lossless could gain and be used mainstream, Wireless earphones are replacing traditional earphones. Meaning your music will be re-encoded before it is sent to your earphone. And No. Most Android or iPhone dont have AAC pass through. i.e Your AAC encoded files will still be re-encoded before sending it your bluetooth earphone.

f33d5173• 3 years ago

I think its worth repeating that "cd quality" is a term of art being used here that does not necessarily mean the audio has the quality of a cd (and, as they emphasize, in fact does not). I would dispute that any new standard has taken the helm of "cd quality" - my experience is that such a phrase is never used in describing quality of lossy compression. Most music downloads are either described by their bitrate (so that the listener is left to figure out what these mean), or by labels like "low", "medium", and "high" quality (with the listener left to distinguish whether those are accurate descriptors).

dang2• 3 years ago

Per Guidelines you should put years in the Title.

It's certainly the convention on HN to put the year in the title for older articles, but it's not one of the guidelines (https://news.ycombinator.com/newsguidelines.html).

(minor point but I can't help it)

_Algernon_2• 3 years ago

Conventions are just guidelines that aren't written down. Also couldn't help it.

jakemauer• 3 years ago

Musepack! There are dozens of us! Dozens!!

Back in the early 2000's when I was getting into ripping my collection I didn't have enough space for FLAC so I surveyed the options and Musepack seemed like the obvious lossy codec winner. I still have that collection of .mpc's somewhere.

xeromal2• 3 years ago

Per

taskforcegemini• 3 years ago

some people encode their mp3s to FLAC

thefilmore3• 3 years ago

I came across this after listening to several 320kbps MP3 files and found that they sounded noticeably worse than 256kbps AAC versions. Support for AAC is widespread now and it should be preferred over MP3 [1].

[1] https://www.iis.fraunhofer.de/en/ff/amm/consumer-electronics...

wooptoo2• 3 years ago

MP3 does have an advantage though, being widespread and royalty-free since the Fraunhofer patents expired.

snvzz2• 3 years ago

OPUS is still better than AAC and MP3.

kalleboo• 3 years ago

AAC-LC patents have also expired since the codec was introduced in 1997 (26 years ago)

olivierestsage• 3 years ago

While acknowledging that I don't know whether I can tell the difference in every case or not, I would summarize my own preference for lossless audio in the following terms. Choosing lossy audio, my best case scenario is that I save space or bandwidth because I can't tell the difference; my worst case scenario is that I'm missing some element of the music, whether it is consciously noticeable, something I'm unaware of entirely, or perhaps something that I may only be experiencing on a somatic level that doesn't reach the level of conscious thought (I know that the possibility of this last option will be contested by some, and that's fair enough). Choosing lossless audio, my best case scenario is that I'm hearing the music in a higher fidelity, and increasing the amount I'm capable of appreciating; my worst case scenario is that I'm wasting some space or bandwidth for the reassurance. Basically, Pascal's Wager, but for audio.

willis9363• 3 years ago

There are measurably much larger effects from insufficient replication hardware. Are you using the same amp, speakers, room, listening position, and volume level as the person who mastered the recording? No? Then your difference in setup is adding much larger differences than -90 dB RMSE.

It's all a painfully fruitless effort when you learn that most masters don't even consider the phasing of instrument microphones and none of it is at all a close approximation of what it would be like to be in a room listening to instruments. It's good enough, yeah, but there are much more important and difficult threads to tug than lowering noise in the signal chain.

denton-scratch2• 3 years ago

I'm convinced that we can "hear" frequencies well above the reputed 20KHz limit of human hearing, as overtones, i.e. as tonal quality.

I certainly don't have golden ears; I'm no audiophile, and I'm getting on in years. 44KHz FLAC is easily good enough for me. But I tire of listening to MP3 music, after a few tens of minutes; it seems to lack the presence and immediacy that keeps me interested.

willis9363• 3 years ago

That's fine, but you should appreciate that you are lying to yourself until you perform a blind test to prove it.

denton-scratch2• 3 years ago

you are lying to yourself until [...]

Not really. I'm not proposing a hypothesis that needs testing; I'm just reporting subjective anecdata. I don't need to test it, because even if I'm deluded it costs me 300GB instead of 100GB. Pfft.

"Lying to yourself" is silly talk; that implies that I'm knowingly telling myself a falsehood, which doesn't make sense. At worst, I'm mistaken.

danbee4• 3 years ago

If frequencies abover 20Khz are causing audible differences in the audible frequency range then that's considered distortion and there's an issue in your playback chain somewhere.

https://people.xiph.org/~xiphmont/demo/neil-young.html

kwanbix• 3 years ago

For a while, HydrongenAudio, arguably the best sound/music related forum, did plenty of listening tests.

The last one sadly is from 2014: they tested Opus, AAC and Ogg Vorbis at 96 kbps against a classic MP3 128 kbps, and find out which codec produces the best sound quality.

https://listening-test.coresv.net/results.htm

https://listening-test.coresv.net/bytrack/index.htm

Notice that it is almost 10 years old, and that MP3 was encoded at 128kbps.

mixmastamyk2• 3 years ago

Interesting—have been tinkering in this area for decades now and always heard AAC was better than MP3. But until now have not seen how/why it was better. Thank you Stereophile.

Yes as several have written, the piece is from 2008 and it doesn't matter any more.

First, once LAME and VBR came about, I've never been able to tell the difference between my 192K MP3 and lossless files, even as a spring-chicken with expensive equipment. Been "good enough" for a very long time.

Second, since storage and bandwidth exploded I've used FLAC exclusively. Why not? But, have found 24/96+ files on the internet occasionally and first thing I downsample them to 16/48khz and do a listening test. I sure as hell can't hear the difference between those. I do leave the last extra 3.9khz... why not? Incredibly cheap and maybe the kids can hear it. Playable on car stereo and more compact, one third the size.

Finally, a big exception. Techies obsess about compression formats, but they don't matter as much as you think at the high-quality end. I've learned the source, i.e. master recording is more important. Example—rip "pristine" FLACs (or WAVs) directly from an iconic 80s CD. Do a listening test. Compare them with a modern remaster encoded with 192K Lame VBR MP3. The MP3 will sound a lot better and preserve the improved high end details. Yes, more noise but you'll struggle to hear it.

(Caveat—this is assuming we're not talking about a shitty 2010-era "loudness war" remaster but a quality-oriented remaster.)

Was mildly surprised by this after insisting on FLAC for almost two decades. A bit too early, in hindsight. Storage is so cheap now though, it again doesn't matter. FLAC it is, Opus from online sources.

fladd• 3 years ago

The intersection of not understanding digital audio and not understanding the neuroscience of hearing remains a place that never ceases to amaze me.

eqvinox2• 3 years ago

The real issue nowadays is that you shouldn't record or master at 44.1kHz (or multiples) and go for 48kHz instead.

But not because it sounds better. Simply because 48kHz is what your computer and phone natively clocks its audio codecs at. It's done because that's generally an integer fraction, but the why doesn't matter as much as the fact itself; PC "HD Audio" and phone codecs are 48kHz.

Yes you can resample, and yes you can resample without it being audibly noticeable. But it's an extra step where you're at the mercy of whoever implements it to do it right. Doing it wrong may also include noticeable delay, breaking e.g. A-V / lipsync.

Can you build HiFi systems that support 44.1kHz and maybe dynamically switch their clock source as needed? Sure. But what's the easiest way to build a HiFi system these days? You just stick an off-the-shelf embedded device in it, which likely uses standard PC/phone tech…

So just ship 48kHz.

(Similar argument for video recording in 50Hz countries btw - unless you are recording for TV/broadcast, you should always shoot at 60 / 59.94fps. Because that's what PC and phone screens run at…)

rplst812• 3 years ago

This used to be somewhat true, but modern digital often uses such high internal rates that converting between the two is near lossless. For instance, the least common multiple of 44.1k and 48k is in the 7MHz region. Realtime conversion between them is pretty doable today. We've been doing it by using decimation with much simpler hardware for a few decades before that.

eqvinox2• 3 years ago

That might be true for the hardware, but since mixing of multiple streams is a software function nowadays (… some older HW was multi-stream capable…), any 44.1kHz stream has a good chance of being resampled to 48kHz just to allow mixing it with other sounds.

Even if it's the only stream and you could switch the codec to 44.1kHz mode, what do you do if the OS wants to play a random notification sound? Switching between 44.1kHz and 48kHz is not going to be hitless on a significant number of HW (not all, but most I'd guess), so whoever's writing your OS mixer code would reasonably make a call to always mix at 48kHz…

(Yes this argument primarily applies to PCs and phones, hopefully on a HiFi system that just happens to use COTS embedded devices they'd write some code to switch the rate…)

8bitsrule• 3 years ago

compressed ilk do not offer sufficient audio quality for serious music listening.

Old old OLD argument here. Apart from -rare-, well-trained golden ears, very few humans can distinguish 192K or better, well-encoded MP3s without knowing -exactly- what to listen for. Or will need to 99% of the time.

As dozens of studies have shown over several decades. The rest is either marketing or self-deception.

https://wiki.hydrogenaud.io/index.php

RockRobotRock2• 3 years ago

Wow, great tests! Now do a blind A/B with headphones on. :)

thefounder• 3 years ago

In the age of 4K,8K video streaming, free efficient lossless codecs and yearly $1000 phone upgrades why do we even bother to ask the question of lossy vs lossless audio for entertainment? It's not a matter of disc capacity anymore. You can have the best for "free".

If we do ABX test you will find out that people can't even make the difference between the original artist and a cover artist let alone lossy vs lossless. Should we just use cover artists at concerts?

The brain adapts quickly to lower quality be it visual, audio, olfaction or gustatory. Does it mean we should ingest the most we can tolerate because we get used to it so we can run on more efficient/cheap resources/content ?

conradfr2• 3 years ago

The whole world does not have FTTH and unlimited 4G+ mobile data plans.

And you would be surprised by the number of radios streaming at 96kbps.

ofslidingfeet• 3 years ago

There are people who can hear up to 22khz and beyond, so I've never understood why many of these arguments aren't easily resolvable on that basis alone.

dusted• 3 years ago

This can be humbling http://abx.digitalfeed.net/list.html just try flac vs 128kbit.. I swear I used to be able to distinguish 320kbit, now I struggle with 96k :s Age and loud music...

MP3 vs. AAC vs. FLAC vs. CD (2008)

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