[HN Gopher] MP3 vs. AAC vs. FLAC vs. CD (2008)
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MP3 vs. AAC vs. FLAC vs. CD (2008)
Author : thefilmore
Score : 89 points
Date : 2023-10-01 14:41 UTC (8 hours ago)
(HTM) web link (www.stereophile.com)
(TXT) w3m dump (www.stereophile.com)
| marcodiego wrote:
| 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 wrote:
| 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.
| thefilmore wrote:
| MP3 smearing can be noticed even at 256kbps [1].
|
| [1] https://www.soundonsound.com/techniques/what-data-
| compressio...
| torrance wrote:
| 2008.
| AshamedCaptain wrote:
| 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...
| isykt wrote:
| 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.
| tamimio wrote:
| 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.
| tjoff wrote:
| 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).
| oriolid wrote:
| 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.
| masklinn wrote:
| 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.
| tamimio wrote:
| > EQ-ing and mastering could also be different.
|
| Possible, although I didn't change any default ones.
| teolandon wrote:
| There is no lossless on Spotify.
| tamimio wrote:
| Well, than that's definitely the reason why
| Scoring6931 wrote:
| It could be simply a difference in sound volume. Our brains
| tend to believe louder is better.
| tamimio wrote:
| 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.
| thefilmore wrote:
| 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...
| wooptoo wrote:
| MP3 does have an advantage though, being widespread and
| royalty-free since the Fraunhofer patents expired.
| snvzz wrote:
| OPUS is still better than AAC and MP3.
| wooptoo wrote:
| I don't disagree. But almost any electronic device released
| in the last 10 years can play mp3. Probably even your
| electric toothbrush.
| CharlesW wrote:
| Even if you set aside universality as a desirable property,
| according to the official site at 128kbps any Opus
| efficiency advantage disappears. https://opus-
| codec.org/comparison/
| brewdad wrote:
| Which is why I use Opus128 on my iPhone, where storage is
| a limited, fixed commodity and my listening environment
| is rarely optimized. Everywhere else is FLAC or mp3-256.
| My ears aren't golden enough anymore to justify 320 bit
| mp3.
| drupe wrote:
| No, you can't hear the difference, let's collectively move on,
| please.
| mjlee wrote:
| 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.
| swagempire wrote:
| 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.
| c0pium wrote:
| No you can't. Maybe it's volume, maybe it's the non-blinded
| nature of the testing, but you can't hear it. They're the
| same.
| amluto wrote:
| ...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 wrote:
| 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.
| throwaway167 wrote:
| 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 wrote:
| Citing his own past work from 1995, too.
| esafak wrote:
| I'm glad it's an old article. It would be sad if those
| audiophiles were still debating this old chestnut.
| Tsiklon wrote:
| 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
| esafak wrote:
| Hydrogen Audio?
| Godel_unicode wrote:
| Plenty of those people are present in this thread.
| raphlinus wrote:
| 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 wrote:
| 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.
| raphlinus wrote:
| 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.
| eviks wrote:
| What is the distortion if no human is able to hear it?
| raphlinus wrote:
| Just to give one example, if you want to do forensic
| analysis on the signal based on inaudible differences.
| That's a valid use case for an archive that doesn't apply
| to consumer (even audiophile) consumption of music
| streams.
| bobsmooth wrote:
| >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.
| RIMR wrote:
| 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.
| snvzz wrote:
| >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.
| oriolid wrote:
| 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.
| 20after4 wrote:
| everyone should really watch https://xiph.org/video/vid1.shtml
| and https://xiph.org/video/vid2.shtml
| [deleted]
| Waterluvian wrote:
| 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.
| conradfr wrote:
| 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...
| jwatzman wrote:
| 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.
| mjlee wrote:
| > 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?
| [deleted]
| ptx wrote:
| 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."_
| eric__cartman wrote:
| 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.
| mrob wrote:
| 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.
| croes wrote:
| Maybe decoding speed is an issue or decoder quality
| ptx wrote:
| 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".
| wesapien wrote:
| 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 wrote:
| You can easily create blind tests and they speak a very clear
| outcome.
| kwanbix wrote:
| 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.
| kristofferR wrote:
| 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.
| eviks wrote:
| 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?
| amluto wrote:
| Have you contemplated what AWS charges to egress a music file
| in 2023? Gigabytes are _expensive_.
| kristofferR wrote:
| I don't see the relevance to AWS regarding people's music
| habits.
| throwaway167 wrote:
| The problem with AWS is the temptation to use lambdas when
| streaming will keep your program running. Adding on egress
| costs, as parent points out, makes what could be a simple
| streaming server quite expensive. Better to avoid AWS for
| this reason. Personally I've had great success with my own
| streaming server I made a few years to learn Rust running
| on a raspberry pi that I leave plugged in at home.
| tjoff wrote:
| The bigger question is why people uses AWS for sending bulk
| data.
|
| Gigabytes are cheap.
| almatabata wrote:
| 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.
| mixmastamyk wrote:
| 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 wrote:
| 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.
| jrajav wrote:
| 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.
| bayindirh wrote:
| 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.
| liquidise wrote:
| 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.
| bayindirh wrote:
| I think you need to spend more time with the systems and
| the music you have. Because, at least for me, understanding
| the differences at the first shot is very unlikely.
|
| Brain is interested in the low hanging fruit, i.e. the
| music and the melody itself, first. The music needs to
| became mundane or ordinary to be able to listen it deeper
| for more details. This is when differences can be heard
| more easily.
|
| Lastly, you don't need perfect systems to hear differences,
| but understand how your systems respond to the music you're
| listening to. i.e., your music system's sound needs to be
| mundane to your brain too to be able to go from low hanging
| fruit to minute differences you were not able to hear
| before.
| duped wrote:
| 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.
| bayindirh wrote:
| This is the first reaction I always get: "This is snake
| oil, and you don't know what you're talking about". While
| there are undisputed snake oils in audio and audiophile
| market, the difference I'm talking about, is not.
|
| The idea of lossy codecs is they filter out the things you
| theoretically don't hear, yes. However, the presumption
| that you don't hear these when they are present is not
| completely true. Because they have a secondary order
| effects in overall sound.
|
| The audible residue you claim that I don't hear when it's
| in the CD is the part of the sound which adds this
| instrument separation and soundstage expansion. Same is for
| higher sampling rates. While you can't pinpoint the
| difference with words, it shows itself as smoothness and
| "richer" sound.
|
| Saying that you can't hear that difference is akin to
| saying "Human eye can't see faster than 30/60/X FPS
| anyways", which is not true.
|
| When anyone presented with a lossy-encoded audio file
| produced with a state of the art encoder and not brick-wall
| mastered, will be impressed, yes. This includes me, too.
| However whenever you listen to the same file in lossless
| or, if present, higher resolution formats, with a
| sufficiently transparent audio system a couple of times,
| you start to notice the differences.
|
| There are a couple of caveats in all of this audio
| business. First of all, you need to know how your audio
| system sounds and behaves to be able to discern
| differences. This requires time with the same system for a
| long time, to understand how it responds. In my case, I
| have the luck of having the same amplifier (An AKAI
| AM-2850) for ~30 years. I know how that thing responds to
| any genre of music, and I know how anything should sound at
| any quality level. Again, as I aforementioned, you need to
| do these ABX tests a couple of times back to back, esp. if
| you don't know the track, to be able to decode the details
| in sufficient manner. Digitalfeed's ABX test
| (https://abx.digitalfeed.net) understands this and makes
| you listen to the same thing 5-10 times according to your
| available time.
|
| See, I'm an ex-orchestra player. I played in concerts,
| listened master recordings, and YouTube uploads of the
| concerts I played as well. I have also listened tons of
| CDs, MP3s of the same albums, etc. Some of the albums I
| listen have a captivating sound when I listen to them from
| CDs. MP3 versions of the same albums do not nail me to my
| chair, yet I can't leave the CD version of the same album
| to get a cup of tea. Both are ran through a Yamaha CD-S300
| CD player with an iPod interface and MP3 playing capability
| over USB.
|
| I can also write how CD tracking quality affects audio
| clarity, but this comment is long enough. In short,
| Yamaha's old CD-Recorder, CRW-F1 really improved sound
| quality by abusing Red Book standard by lengthening the
| pits of audio CDs. It reduced to capacity to 68 minutes,
| but it was worth it, esp. on lower end CD players.
| mmastrac wrote:
| > I can also write how CD tracking quality affects audio
| clarity, but this comment is long enough. In short,
| Yamaha's old CD-Recorder, CRW-F1 really improved sound
| quality by abusing Red Book standard by lengthening the
| pits of audio CDs. It reduced to capacity to 68 minutes,
| but it was worth it, esp. on lower end CD players.
|
| Sorry this one part especially makes no sense. Digital is
| digital. Either it added more samples per second, or more
| bits per sample, or it's snake oil. There's a stream of
| bits that comes out of the reader. There's no residual
| information about the length of the pits.
|
| EDIT, yeah sorry this is completely and utterly
| impossible that you are getting better "audio clarity":
|
| "Yamaha tries to attract computer enabled audiophiles
| with the Audio Master technology. Audio Master promises
| reduced jitter and decreased error rates for audio
| recordings via extended pit and gap sizes on the CD-R.
| This is actually quite simply achieved by increasing the
| disc rotation speed vs. the laser clock frequency. In
| other words, Audio Master recording at 8x rotates the
| disc at 8.2x, thus creating the extended pit & gap
| lengths. This naturally reduces the capacity of the
| disc."
|
| Literally they are just spinning the disc faster,
| reducing capacity and make it slightly less likely that
| errors will be read. If you're getting read errors on
| playback, that means your disc is dirty or your CD player
| sucks. It's the same bitstream, just read at a different
| linear rate.
|
| If you honestly believe that this is an audiophile
| concern, I'd urge you to reevaluate a lot of your other
| beliefs, because they are clearly not all grounded in
| technical facts.
| bayindirh wrote:
| CRW-F1 didn't encode more information into the pits. It
| allowed lower end DACs to have more time to switch
| properly by giving them a slower signal stream within
| acceptable limits.
|
| DAC's digital part is easy. What differs in quality is
| the analog part. If DACs were that simple, a 25 cent DAC
| would power every unit from bottom bin to top tier.
|
| Before that Yamaha CD player I had, I used a lower end
| Sony CD-Player (I don't remember the model, sorry).
| Writing the same album, to same brand of CD-R, with the
| same speed in two different modes created two audibly
| different disks.
|
| I sometimes challenged myself by writing in both modes,
| not marking the CD-Rs, and the audio difference was
| always audible. Even after weeks. 68 minute CDs were
| always had larger sound stages with more clarity and
| instrument separation. This is again on the same AKAI
| AM-2850 amplifier.
|
| I guess this difference would be impossible to hear
| today, because higher end units have better tracking and
| better DACs. Also some of them use DAE and use multi-
| second buffers, so the "slower stream" is no longer
| present in the pipeline due to buffering.
| mmastrac wrote:
| A CD signal stream fed to a DAC is a 44.1 kHz 16-bit
| signal, period. All you did was force the drive to spin a
| bit faster to keep tracking, or let it fill its buffers
| more slowly if it spun at the same speed. The buffers,
| after error recovery, are what feed the DAC. Assuming an
| error-free read on two discs burned with the same data
| (regardless of "pit length", disc material, etc), you get
| the same bits in the buffers.
|
| There's no "slower bitstream" for the DAC. That's
| provably nonsense and you can work it out from basic
| principals. The same bits would come out of the optical
| interface of a CD player, at the same rate either way. If
| the CD player has a built-in DAC, the same bits would get
| fed to that same DAC either way.
|
| I'm sorry, but if this is truly what you believe, it
| really puts everything else that you said into question.
|
| To give you the benefit of the doubt, I might say that
| the lenses or lasers on your CD players are filthy, and
| you're just hearing skips or noise from poor reads and
| that a slower-written, borderline-spec disk might just
| allow them the function better. Perhaps your player was
| interpolating or concealing frames [1] that it couldn't
| read correctly and failed to correct via ECC and you were
| just hearing a poorly reconstructed digital data stream.
|
| This sort of confident incorrectness, ignoring the
| underlying technical architecture, is probably why people
| don't believe anything that an audiophile says.
|
| [1] https://www.pearl-
| hifi.com/06_Lit_Archive/02_PEARL_Arch/Vol_...
| Syzygies wrote:
| "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.
| mrob wrote:
| >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.
| Syzygies wrote:
| Ok, so despite anecdotal evidence that some individuals can
| distinguish better-than-CD quality audio, we're questioning
| the existence of convincing double-blind studies. Yet we
| accept the varying cutoffs for what frequencies a person
| can consciously detect in isolation, as proof that we are
| incapable of perceiving audio information above those
| frequencies.
|
| Are people asserting that an ear removed from a cadaver,
| hooked up to the best available scientific equipment,
| measures as a perfect biologically derived low pass filter?
| Or that we even partially understand how neurons work, when
| there may be quantum effects to be uncovered a century from
| now?
|
| Intellectual history is a graveyard of models confused with
| reality.
| beebeepka wrote:
| "The human eyes can only do 24 fps" and "if I can't do thing,
| then no one can" all over again.
| zepolen wrote:
| No one ever said the human eyes can only do 24fps.
| TylerE wrote:
| No one who knows what they're talking about, but I've
| absolutely seen that argument advanced on multiple
| occasions, albeit usually with 30 rather than 24fps.
| beebeepka wrote:
| Last time I heard this crap in real life was back in
| 2018. Things have progressed immensely over the last
| decade, though. Now almost everyone "knows better" due to
| relentless marketing from big companies, including phone
| and TV vendors.
|
| Before that, 30 was totally fine for the masses. In fact
| it was preferable. Cinema is the last big holdout and,
| apparently, it's going to take at least another decade
| before even mere 48 is standard. As someone who has been
| riding the 120+ fos for over two decades, going to the
| movies is awful, especially action scenes and panning.
| throwaway54_56 wrote:
| 24 fps is the standard for cinema because that gives the
| preferred look for most content, with nice looking motion
| blur and whatnot. High frame rate may make sense for some
| movies, but it's not a win for the whole industry to go
| 48 or 60 or higher.
|
| I'm not sure what you're getting at with the 120 fps
| comment, because that is obviously not the frame rate of
| the finished product, so it's not the same conversation.
| beebeepka wrote:
| The whole 24 FPS thing is mostly historical. I don't like
| it. Nor do I like motion blur.
|
| The 120 fps was regarding games. While movies are
| passive, they could still benefit immensely by doubling
| to 48. Not every scene in a movie is people talking and
| this is where 24 stops being adequate. Even YouTube has
| had support for 60 FPS videos for years.
|
| I know it's not a win for the movie industry. They ought
| to hate it, especially the artsy types.
| TerrifiedMouse wrote:
| > Cinema is the last big holdout
|
| Don't know if cinema will ever drop 24fps. The shift to
| higher frame rates is of questionable benefit as it just
| makes movies look like TV shows. It seems 24fps is what
| makes a movie feel like a movie.
| ScoobleDoodle wrote:
| What quality and power speakers are needed to get good output
| from the files so it can be heard?
| nix0n wrote:
| > 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...
| saaaaaam wrote:
| 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.
| zigzag312 wrote:
| > asked people to tell us which was higher or lower quality
|
| This test didn't measure what you probably wanted it to
| measure.
| lmm wrote:
| 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").
| tzs wrote:
| 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
| asveikau wrote:
| > 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.
| ok_computer wrote:
| 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.
| ok_computer wrote:
| Also, I'll note that I think the amp and speakers are far
| more contributing than the master file format. And the
| quality of the master and mix and tracking even moreso.
|
| I'll run youtube rips of dj sets through some light
| hardware compressors and preamps and it sounds great. You
| cannot have specs determine quality.
| Blackthorn wrote:
| Yes, that's for antialiasing headroom purposes during the
| production process.
| SSLy wrote:
| that's what 'production' in the quoted passage means.
| daneel_w wrote:
| 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.
| vel0city wrote:
| > 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.
| masfuerte wrote:
| It's perfectly preserved only if your samples are perfect.
| Imagine instead that we used 4-bit samples. The results
| would be obviously garbage. 8-bit would be better. 16-bit
| is better still. But it isn't perfect.
| daneel_w wrote:
| I doesn't look like you understand what sampling is, and
| how reconstruction filters in DACs work. Your statement is
| true for _some_ waveforms, depending on their frequency,
| due to the use of reconstruction filters on the output, but
| it 's not true for any signal and the problem becomes more
| apparent the higher the frequency of the waveform.
| krackers wrote:
| If I'm understanding you correctly, you're saying that
| while a perfect sinc interpolation reconstruction would
| allow you to capture up to 44.1/2 kHz, in practice since
| we're limited to FIR reconstruction filters we can't
| actually get that high? If so it seems like a fair point,
| although I'd imagine they'd be better than 6khz?
|
| There's also the issue of the input signal not being
| band-limited which is necessarily true for real world
| signals given that you sample for a finite duration.
| [deleted]
| [deleted]
| ReactiveJelly wrote:
| Yeah this is the "stairstep vs. lollipop" thing again.
| kevin_thibedeau wrote:
| Everyone complaining with "but stairstepping" fails to
| recognize that the final stage of a DAC is a reconstruction
| filter. The steps are gone after that filter is applied.
| You aren't analyzing the full DAC performance if you look
| in front of the filter. This is most dramatic in class-D
| amplifiers where the raw waveform feeding into the speakers
| is square wave hash that gets filtered out by the speakers
| themselves.
| daneel_w wrote:
| The filters do linear interpolation between samples. This
| bridges some shortcomings of a sample rate too low to
| capture complex waveform at high frequency, but it's not
| a silver bullet.
| daneel_w wrote:
| Yes and no. Reconstruction filters are part of the problem
| (and part of the solution) but it's not all about them.
| eyegor wrote:
| 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
| daneel_w wrote:
| A 44 kHz sample rate guarantees accurate 22 kHz _triangle
| wave_ reproduction if a reconstruction filter with linear
| interpolation is used on the output, and accurate amplitude
| of same signal if samples happen to align somewhat with the
| peaks of the waveform.
| user_7832 wrote:
| Yep, as Monty showed in the 2nd xiph video a square wave
| will have issues with a low nquist frequency (at for eg
| 44khz sampling).
| pja wrote:
| 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.
| user_7832 wrote:
| > https://xiph.org/video/
|
| Thanks a lot for those videos, they were absolutely
| excellent. For anyone wondering they're presented by
| "Monty", the guy behind the ogg container and vorbis codec.
| I probably understood 10% of what he said but that's still
| a lot.
| temp0826 wrote:
| 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
| daneel_w wrote:
| It's true for a triangle wave and a square wave, depending
| on if the output has a reconstruction filter doing linear
| interpolations between samples. You cannot accurately
| sample a 22.05 kHz sine wave (or any other "complex"
| waveform) with a 44.1 kHz sample rate.
| dghughes wrote:
| > 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?
| brewdad wrote:
| On $10 cheap buds? Yes. On $100 middling buds? Very, very few
| people will notice a difference.
| ptx wrote:
| 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.
| criddell wrote:
| 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?
| CharlesW wrote:
| > _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.
| sircastor wrote:
| 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.
| mrob wrote:
| 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).
| elzbardico wrote:
| The same can be said about loudspeakers, a good set of
| loudspeakers is far more important than buying a super-
| expensive set of DACs, Pre and Power Amps.
| crazygringo wrote:
| > _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.)
| jart wrote:
| People talk about dynamic range compression all the time.
| kevin_thibedeau wrote:
| 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.
| jrajav wrote:
| 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.
| amluto wrote:
| I haven't done the math, but I wouldn't be utterly shocked
| if undithered 16-bit audio, cranked up some silly amount
| (such that full scale is 130dBA perhaps) has an audible
| noise floor.
|
| This is consistent with my other comment about _badly
| encoded_ MP3 being far from transparent.
| SSLy wrote:
| Yeah, 18-20 bits make sense in loudly tuned cinemas.
| eviks wrote:
| "the effective dynamic range of 16 bit audio reaches 120dB
| in practice" https://people.xiph.org/~xiphmont/demo/neil-
| young.html#toc_1...
| lmm wrote:
| That analysis misses that when you dither you sacrifice
| effective sampling frequency for dynamic range.
| 44.1KHz/16bit can represent that dynamic range, but it
| can't represent that dynamic range at a 44.1KHz sample
| rate.
| goalieca wrote:
| > 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.
| mrob wrote:
| 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
| joshspankit wrote:
| I wish years ago we would have switched to A/B testing taking
| weeks or months for each side.
| dwroberts wrote:
| 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.
| lmm wrote:
| 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.
| mrob wrote:
| 48kHz has the advantage of being an integer multiple of many
| common video frame rates, which makes video editing simpler.
| soulofmischief wrote:
| 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.
| pja wrote:
| 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.
| kevin_thibedeau wrote:
| Even 128kit from a modern encoder is harder to pick out
| than it was 25 years ago. Most of the self-appointed
| experts proclaiming how woefully inadequate MP3 is are
| basing their assessment on outdated experience from the
| distant past.
| lmm wrote:
| > 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).
|
| AIUI there are some things that don't go away at any bit-
| rate, e.g. pre-echo.
| c0pium wrote:
| 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.
| soulofmischief wrote:
| Maybe not anyone. I could be biased with sensitive hearing.
|
| I know that in grade school, part of the requirements for
| joining band, due to the size of my school and overwhelming
| demand, was passing an audiometry test, where we were
| evaluated on a few different contexts related to ability to
| discern detail in audio, such as pitch and volume. I
| remember being pulled away into the principal's office
| where some of the test administrators were present, and
| they accused me of cheating and demanded to know how I did
| it.
|
| Apparently, I was the only student in the entire state to
| get a perfect score on that test, at least for that
| particular year. Unsure if they were implying I was the
| first ever, but that seems ridiculous to me because passing
| the test boiled down to just paying close attention.
|
| So I really don't know. Maybe the average person can't hear
| it, but I know just what to look for in the high-end and
| usually guess even 320kbps mp3 correctly from my own self-
| tests wherein I would randomly select between different
| encodings of a music file. I'm confident I would do well in
| an administered ABX test if I'm simply being tasked with
| finding the difference between mp3 lossy encodings and a
| lossless reference.
| MaxBarraclough wrote:
| I'm not sure I get you. If someone can't tell the
| difference in a blind test, that means they can't tell the
| difference. The result of a non-blinded test is of no
| consequence.
| m463 wrote:
| 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.
| Slow_Hand wrote:
| 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).
| Slow_Hand wrote:
| Correction: In the last paragraph of my comment I mistakenly
| typed "latency artifacts" when I should have said "aliasing
| artifacts".
| ReactiveJelly wrote:
| 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?
| [deleted]
| matsemann wrote:
| The point is that because of sampling, order of operations
| can matter. So having a 88k file -> apply an effect ->
| downsample to 44k, can sound different than having a 88k
| file -> downsample to 44k -> apply an effect.
| S_A_P wrote:
| This is an important point. The main reason that pro
| audio gear pushes bit depth and sample rate up to higher
| that 16/44.1 audio is because when you start doing the
| floating point math to mix and apply effects to audio you
| can end up with audible differences when multitrack
| recording. In this case (and I still think it's optional
| for all but the most demanding recording of live
| performance) higher sample rates can help and to a lesser
| degree but depth can give you more dynamic range.
|
| I give that long preamble to say once a record is done
| and mastered, having > 16/44.1khz is wasted bandwidth.
| junon wrote:
| You can verify this by mixing to mono or splitting stereo
| and inverting the "after" and mixing them back into the
| "before".
|
| If you get silence, they're perfectly identical.
| Blackthorn wrote:
| The downsampled 44k that went through a half rate filter
| might actually sound better, for that matter. The speakers
| won't try to reproduce the content above 22khz then.
| high_priest wrote:
| If you think about this "aliasing" as in, what occurs in 3d
| graphics, then you can understand this. What these 3d
| fiters do is either remove infirmation with blur (FXAA) or
| use information that is not available in the image (MSAA
| and derivatives)
|
| In audio recording, sampling at 88k would be like
| generating MSAA x2 image, so it can be displayed with higer
| fidelity, despite the outgut resolution being in lower 44k
| sampling rate.
| pkulak wrote:
| 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.
| mattgreenrocks wrote:
| 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.
| pimeys wrote:
| And 128k opus is perfect to my ears. I store all my music
| in the best possible quality FLAC files, but stream it to
| my phone in 128k opus. Such a great format, encodes very
| fast even with my Intel atom and sounds great.
| Espressosaurus wrote:
| 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.
| c0pium wrote:
| I have very good equipment and lots of people in my circle
| who are audio enthusiasts. None of them have ever been able
| to demonstrate in a blind test that they can tell the
| difference.
| jerf wrote:
| 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
| TylerE wrote:
| The giveaway for low-mid nitrate MP3 is the high hats. The
| lower the nitrate, the more you get a sort of temporal
| ghosting that sounds like an almost "crunchy" swishy sizzle
| sort of sound, a bit like a jazz player using brushes, but
| more lo fi.
| hunter2_ wrote:
| I agree, and my hypothesis is that it's exacerbated by
| the combination of three particular things:
|
| 1. It's a high frequency complex waveform with a fast
| envelope, so it demands bitrate.
|
| 2. Drum miking often involves multiple mics spaced apart,
| so more than one typically picks up any given cymbal with
| a phase offset, and those mics are panned quite
| differently, leading to a very "wide" result, i.e., left
| and right output is fairly uncorrelated as seen on a
| vectorscope [0].
|
| 3. A perceptual codec at a given total bitrate often
| sounds better when stored as a mid-side transformation
| (instead of storing a left channel and a right channel,
| store a L+R "mid" a.k.a. sum channel and a L-R "side"
| a.k.a. difference channel), also known as "joint stereo"
| which is a common flag on MP3 encoders, because it allows
| for assigning more bits to the mid channel (correlated
| signals) and fewer bits to the side channel (uncorrelated
| signals). More bits for mono center-panned stuff like
| vocals is the goal, which is generally for the best, but
| fewer bits remain available for wide stuff like those
| cymbals! Contrast with regular stereo mode where half of
| the total bitrate is assigned to each channel. MP3 below
| 256kbps typically needs joint stereo mode enabled in
| order to sound decent.
|
| [0] https://en.m.wikipedia.org/wiki/Vectorscope#Audio
| tedunangst wrote:
| Low nitrate MP3 is a fantastic typo.
| musicale wrote:
| I was going to say this - cymbals are often very
| noticeably bad on MP3 recordings.
| ahofmann wrote:
| Well, most classical songs are very well compressible,
| because not much is going on. Punk Rock or any other music
| were a lot is happening, at the same time, can suffer very
| audible from 128 kbit lossy compression. So you can hear
| lossy compression better in a loud pop song than other
| music.
| esquivalience wrote:
| I don't agree with this from my own experience. To me,
| classical music at high compression suffers far worse
| than modern bands.
| colejohnson66 wrote:
| My unscientific guess would be that classical music might
| have wider dynamic range than "normal" music. So the same
| compression amount affects the one with more range first
| (classical).
| nuancebydefault wrote:
| Higher dynamic range and typically also more 'pure'. The
| introduced compression artifacts stand out more in
| simpler waveforms than in wavevorms that are an addition
| of many more layers of sound.
| replete wrote:
| 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.
| deaddodo wrote:
| 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.
| nyolfen wrote:
| "you need more than anecdotal evidence"
|
| "have some anecdotal evidence"
| replete wrote:
| [flagged]
| fsckboy wrote:
| things that are slightly louder "sound better". How did you
| control this sort of thing?
| eviks wrote:
| Sure, but that's also easy to normalize in a proper test
| eviks wrote:
| 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...
| jbverschoor wrote:
| We also can't see more than 60 fps according to so much
| research. And why would we want 10 bit screens?
|
| I checked out the link, and the Sample 2 file does not
| represent any wave and is not audible, so the article
| contradicts itself.
| ReactiveJelly wrote:
| We would want 10 bit screens because the research
| indicates that the dynamic range of human vision is
| around 90 dB or 1:1,000,000,000, which is alarmingly
| higher than even 1:1,024
|
| https://en.wikipedia.org/wiki/Dynamic_range#Human_percept
| ion
|
| If all research is wrong, I'm gonna start drinking
| vinegar and building perpetual motion machines :P
| jbverschoor wrote:
| According to Pantone, "Researchers estimate that most
| humans can see around one million different colours". So
| research says we only need 7 bits.
|
| "Research".. sponsored by corporations, and peer-checked
| by scientific voting rings. A bunch of incrowd elitists
| who like to use jargon. Science and politics these days
| are pretty similar
| crthpl wrote:
| The 7 million is probably how many different hues we can
| see. We can see many more different brightness levels.
| Eisenstein wrote:
| Are you both talking about the same thing? Is dynamic
| range the same thing as 'number of colors'?
| chadaustin wrote:
| Where does this "can't see more than 60 fps" rumor come
| from?
|
| It's trivially refutable by placing a 60 Hz strobe (e.g.
| old fluorescent light or even some aftermarket
| headlights) at the corner of your vision.
|
| Also, for interactive systems, 16 ms is a large chunk of
| our reaction time. You need close to 1 ms response times
| (1000 fps) to approximate pen and paper.
| jbverschoor wrote:
| I don't know where it came from.. it was already there in
| the CRT times.
|
| A simple google on 60 fps will still show these
| "scientists" who claim that we can perceive anything
| higher than 30-60 fps.
|
| "Science" does NOT equal truth.
| Eisenstein wrote:
| You seem to be the only one claiming this bit of
| 'science'. No one else has heard of this claim.
| mrob wrote:
| What exactly do you mean by "see more than 60fps"? It's
| possible that 60fps video with full temporal antialiasing
| and low to moderate motion speed could fool untrained
| viewers, but if I'm allowed to move my eyes I can tell
| the difference between high frame rate video (simulated
| with strobing LEDs because of lack of suitable video
| hardware) and real-life motion well into the thousands of
| frames per second. This isn't an unusual ability:
|
| https://journals.sagepub.com/doi/10.1177/1477153512436367
|
| Note that 2kHz flicker requires 4000fps to be displayed
| as video.
| deaddodo wrote:
| I think people are also equating apples to oranges here.
| Vision is analog. There is no "DPI" or "FPS" that human
| vision can see. Some types of motion the human eye can
| perceive at thousands of "frames" and others it can only
| perceive at 60, some colors (green) and contrasts it can
| distinguish extremely fine detail in and other's (blue),
| it cannot. Ultimately it's variable and non-digital so
| it's never going to equate to some strict terms.
|
| The audio, on the other hand, that reaches your ears
| comes _from_ an analog source, even if it ends up digital
| in between. There aren 't some resolution arguments to be
| made here, all that matters is that the output device can
| accurately reproduce the proper analog signal. Which has
| been proven time and time again, and that any
| simplification of said signal is imperceptible to
| anything but the most finely tuned listening devices (or
| maybe some special "golden ears" that the vast majority
| of audiophiles don't belong to).
| user_7832 wrote:
| >> 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.
|
| Correct me if I'm wrong, but isn't 16 bit = 120db about the
| levels of _gradations_ of sound? Even a 4 bit = 16 levels
| of sound pressure /SPL could go from 20db, 20+12.5=32.5db,
| 32.5+12.5db and so on until 120db.
|
| Then, the important question is what's the _minimum_ SPL
| difference perceptable (at a given spl level). That may
| well not be 1db.
| jbverschoor wrote:
| These days "good equipment" unfortunately means:
|
| - Sonos
|
| - Airpods
|
| - Beats
| pimeys wrote:
| All with Bluetooth compression...
|
| For that price range, Hifiman produces pretty good planar
| headphones. The edition XS sounds really good.
| amlib wrote:
| 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.
| Mistletoe wrote:
| 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).
| ReactiveJelly wrote:
| Why AB and not ABX?
| wuiheerfoj wrote:
| Because the base rate is 50% in an either/or test
| eredengrin wrote:
| 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/
| miav wrote:
| 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.
| high_priest wrote:
| 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.
| jrajav wrote:
| 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.
| analog31 wrote:
| 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 wrote:
| 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.
| denton-scratch wrote:
| 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.
| willis936 wrote:
| That's fine, but you should appreciate that you are lying to
| yourself until you perform a blind test to prove it.
| denton-scratch wrote:
| > 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.
| ksec wrote:
| 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.
| dang wrote:
| > 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_ wrote:
| Conventions are just guidelines that aren't written down.
| Also couldn't help it.
| dang wrote:
| Hmm. We scold people for breaking guidelines but we don't
| scold them for not following conventions. We expect
| commenters to know the guidelines but we don't expect them
| to know the conventions. Seems different to me!
| jakemauer wrote:
| 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.
| f33d5173 wrote:
| 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).
| swagempire wrote:
| 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 wrote:
| YouTube is quite good with >128kbps opus audio
| swagempire wrote:
| 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.
| criddell wrote:
| This test seems to imply that it _is_ hi-fi.
|
| https://listening-test.coresv.net/results.htm
| oittaa wrote:
| Not just acceptable. For most people it's basically
| indistinguishable from uncompressed CD audio.
| mixmastamyk wrote:
| 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 wrote:
| The intersection of not understanding digital audio and not
| understanding the neuroscience of hearing remains a place that
| never ceases to amaze me.
| gabereiser wrote:
| 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>.
| flashback2199 wrote:
| For mp3, for me, 192kbps and higher is where it sounds pretty
| good, 128kbps sounds bad
| mrob wrote:
| 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.
| flashback2199 wrote:
| oh no here come the audiophiles _runs away_
| mikeytown2 wrote:
| Didn't use lame for mp3 so the conclusions are pointless in my
| opinion
| thriftwy wrote:
| Lame is the most popular encoder so it is highly likely you
| have listened to its output.
| ck45 wrote:
| 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 wrote:
| Quite literally the sound of rock music is the sound of
| distortion. The kinks didn't razor blade their amps for no
| reason.
| olivierestsage wrote:
| 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.
| willis936 wrote:
| 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.
| lifthrasiir wrote:
| > 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.
| akira2501 wrote:
| > 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.
| jraph wrote:
| 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.)
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