
24/192 music downloads are silly - tosh
http://xiph.org/~xiphmont/demo/neil-young.html
======
errantspark
I once spent ~2 hours explaining all of this (well maybe not all of it) to a
friend of mine who was studying sound design at the time. He didn't believe
me.

I even transcoded some 24/192 FLAC Pink Floyd I had lying around and made him
do a double blind test to show him that he'd prefer the slightly louder song
every time, even if the louder song was 192kbps vs the FLAC. He did. He still
doesn't believe me.

He still thinks he can hear the difference between FLAC and MP3 to this day.
He works as a sound engineer now.

I don't think any amount of reasoning will make some people change their
minds. Some people buy $500 wooden knobs to make their volume pots sound
better. (or was that a hoax? i can't tell anymore)

~~~
masklinn
> Some people buy $500 wooden knobs to make their volume pots sound better.
> (or was that a hoax? i can't tell anymore)

Some people buy small pyramids to elevate their cables off the floor, some
people buy mats to put onto your CDs before putting the CD in a player
([http://dagogo.com/millenniums-m-cd-mat-carbon-cd-damper-
revi...](http://dagogo.com/millenniums-m-cd-mat-carbon-cd-damper-review)),
some people buy $1000+/meter _digital_ interconnect cables
([http://www.theabsolutesound.com/articles/transparent-
referen...](http://www.theabsolutesound.com/articles/transparent-reference-xl-
digital-link/)), some people buy $7200 power cords
([http://www.theabsolutesound.com/articles/crystal-cable-
absol...](http://www.theabsolutesound.com/articles/crystal-cable-absolute-
dream-speaker-cable-interconnect-and-power-cords/?page=5)) and $350/m HDMI
cables ([http://www.theabsolutesound.com/articles/nordost-releases-
fi...](http://www.theabsolutesound.com/articles/nordost-releases-first-
american-made-high-speed-hdmi-cable/)).

Self-styled audiophiles are, by and large, idiots with way too much money
plagued by magical thinking. Developer bullshit has nothing on them.

~~~
Wicher
> some people buy mats to put onto your CDs before putting the CD in a player

An 80-minute, _700_ MB CD-R fits 80 * 60 * 44100 * 2 * 2 / 2^20 ~= _807_ MB of
audio.

Why is that?

The 100MB difference is not just due to the audio TOC being of smaller size
than the ISO9660 or UDF file system metadata. It's also because of differences
in error correction. I don't have the spec on hand but I recall from when I
was investigating this that CD-ROMs use more bits for error correction than
audio CDs. That's why you can fit more audio data than "filesystem data" on a
CD-R. Reading (ripping, digitally) an audio CD will likely result in different
digital audio files every time, since the error correction is not _that_ good,
but good enough, for audio.

I read into this when I was wondering why my CD-DA extracted .wavs came out
with a different checksum every time. Vibration is one of the factors that
would make the same audio CD, read with the same CD player, produce different
digital signals some of the time or even every time.

CD-ROMs however, which store digital data, need better correction - you
definately don't want a bitflip in your .exe, while a minor amplitude diff —
an uncorrected bitflip in the upper bits of a 16-bit PCM signal — is no
biggie.

So… I'm not saying that the people using CD mats are informed (or have tested
whether the mat makes a difference, or would even know how to go on about
testing this, scientifically), but there's more to it than what I originally
thought — which was "it's digital so it's never degraded". I wouldn't have
known without checking the md5sum of my .wav, though.

~~~
Lazlo_Nibble
Uh, no. Bit-perfect ripping is trivial and routine, and tools like the
AccurateRip DB (which has checksums for around three million different titles
you can use to verify the checksums on your own rips) and the CUEtools
database (which has recovery records you can use to correct bit errors on your
own rips) prove it. I routinely get bit-accurate single-pass high-speed rips--
no "paranoid" settings or re-reads--of discs dating back thirty years or more,
and so do hundreds of thousands of other people. If you get different
checksums on successive rips of the same CD, either the disc is damaged or the
drive you're using is failing.

~~~
Stratoscope
Oh sure, your rips may be perfect at the _bit_ level, but how do you know that
they're free of sub-bit quantization that isn't detectable by electronic
circuits but can be heard by the human ear?

This sub-bit jitter and interference can travel along with a digital file and
sneak right past your ordinary bit-level error detection and correction, no
matter how lossless you make it. That's because these errors aren't visible in
the bits. They occur at a deeper and more subtle level, in between the bits.

Even if you prove mathematically that two files contain the exact same bits,
you can't prove that the human ear won't hear any difference, can you?

~~~
the_solution
The decoder/player doesn't know how to read between the bits.

Same file -> same playback.

If you hear the same sound file twice (or two identical files) and hear
something different, you software is broken or you're imagining things.

~~~
Stratoscope
Ah, well, the human ear is a much more finely tuned instrument than your
decoders and players. Think of the feelings you get when you hear the ocean
waves, the birds sing, a thunderclap!

Can you turn this into mere "bits"? Of course not!

That's why it is so important to protect against sub-bit quantization errors,
and this can only be done with proper interconnects. Ordinary cables allow the
bits to travel willy-nilly until they jam up against each other creating a
brittle, edgy soundstage. Quality interconnects are tuned, aligned, and
harmonically shielded to keep those precious bits - and the all-important
spaces between them - in a smooth flow.

And then, we can hear all of the things that make us human.

~~~
eric_h
I'm very glad you stuck with the bit (har har) and didn't resort to just
telling him he missed the joke. Well done.

------
diydsp
I agree with the silliness of 192kHz, but not 24-bits. Here is why:

In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz)
are recorded with lower amplitudes because our ears are more sensitive to
them.

Sampling theory is correct and 16-bits can reproduce any waveform with ~100dB
of range, however, in a complex waveform consisting of low, mid and high
frequencies, the mid- and hi-range frequencies quite simply get shortchanged.

Imagine a recording of a bass sinusoid and a mid-range sinusoid of equal
volume. It might use e.g. 10 bits to store the bass and only 6 to store the
high frequencies. (2^10sin(200wt)+2^6sin(4000wt)). That means the resolution
of the high frequencies is less than the lower frequencies. When the volume of
those frequencies changes dynamically, the high frequencies' amplitudes are
more quantized. That is quite simply why 16-bits are not enough.

This is similar to the problem with storing waveforms unprecompensated on
vinyl. The precompensation makes up for the non-uniformity of the medium. It
could be done with 16-bit digital as well. Or alternatively, larger sample
sizes like 24 can be used.

I haven't A/B tested this. The A/B test in the article compares CD with SACD.
SACD isn't PCM, so its artifacts are going to be totally different from 24-bit
PCM.

~~~
sillysaurus3
The correct way to attack this isn't by attacking the theory. It's to gather a
lot of people and ask them to press a button indicating whether the audio they
hear is 16-bit or 24-bit.

If the results are no better than chance, then 24-bit doesn't matter,
regardless of how sound the underlying argument is.

EDIT: The experiment would also be extremely difficult to design. For example,
you'd need to run this test with music, not simple sounds. So the question is,
which music? I think whatever is most popular at the time would be a good
candidate, because if people are listening to music they hate, they won't care
about the fine details of the audio. But that introduces an element of
uncertainty and noise into the results which is hard to control for.

Some people might deliver accurate results with
[https://www.youtube.com/watch?v=2zNSgSzhBfM](https://www.youtube.com/watch?v=2zNSgSzhBfM)
but not with
[https://www.youtube.com/watch?v=4Tr0otuiQuU](https://www.youtube.com/watch?v=4Tr0otuiQuU)
whereas for others it's the opposite.

Or, it could be the exact opposite: Maybe you can only detect whether a sound
is 24-bit when it's a simple tone, and _not_ music.

Age is also a factor. My hearing is worse than a decade ago.

The headphones used by the test are another factor. If you feed 24-bit input
to headphones, there's no guarantee that the speakers are performing with
24-bit resolution. In fact, this may be the source of most of the confusion in
the debate. I'm not sure how you'd even check whether speakers are physically
moving back and forth "at 24-bit resolution" rather than a 16-bit resolution.

~~~
kazinator
24 bit resolution is important for capture, because it leaves headroom for
mistakes. 16 bits is enough for mastering.

~~~
hackinthebochs
There's also headroom for the signal processing in the equipment. Equalization
or volume control done poorly can lower your dynamic range, for example when
turning the volume down on windows then turning it up on an external amp.

------
the_duck
There's a special irony to the fact that this high fidelity audio format is
being promoted by Neil Young. Young's a rock musician. He's been around loud
noises (e.g. rock concerts) most of his life. He's also 69 years old. Our
ability to hear high frequencies decreases dramatically with age and exposure
[1]. If anyone were able to discriminate 24/192 from 16/44.1, it sure as heck
wouldn't be an elderly rock musician.

[1]: [http://www.patient.co.uk/health/presbyacusis-hearing-loss-
of...](http://www.patient.co.uk/health/presbyacusis-hearing-loss-of-older-
people)

~~~
scarecrowbob
To be fair, I have worked with older sound engineers, and they can hear a lot
of audio artifacts that I miss, just because they've been paying closer
attention for a lot longer than I have, much in the same way my wife (who
plays violin 6+ hours most days) can hear tuning and pitch problems better
than I can.

High frequency limiting is not the only artifact that results from data
compression.

~~~
Johnny_Brahms
Yeah. I worked for a long time as a professional musician in an orchestra. I
fucked up my hands from practising too much, so I switched career.

I can reliably hear a pitch difference of ~0.2hz at this site:
[http://tonometric.com/adaptivepitch/](http://tonometric.com/adaptivepitch/)

and that is after 15 years in a symphony orchestra having my ears blasted by
the brass and percussion section (with a demonstrated hearing impairment from
my time in the orchestra).

------
IvyMike
The industry wants to be able to sell you something "better" and 24/192 is
clearly bigger and therefore better than 16/48.

This is the same reason I'm convinced we're going to get 8k phone displays
someday.

If the recording industry wants to sell me a "platinum" version of recordings,
what I'd really like to have is different mastering of an album: at least one
for noisy environments like the car, and one for higher-quality environments
like my home theater. If you're familiar with "The Loudness Wars", this is a
reaction to that. NiN tried to do this with their "audiophile" mix of
Hesitation Marks (although a lot of people think they did not succeed,
[http://www.metal-fi.com/terrible-lie/](http://www.metal-fi.com/terrible-lie/)
)

On the other hand, I don't need to buy any new equipment to support that, so
the equipment guys aren't going to be happy. I don't know if there's any
silver bullet for them--if there is a hypothetical advancement that would
cause me to upgrade my system, I can't envision it.

~~~
bsder
Until phones are 600dpi like paper, I'm fine with display resolution
continuing to increase, thanks.

~~~
thedufer
4k would exceed that significantly on a 6" display. Thus why parent compares
an 8k phone to the 24/192 discussion - the benefits are nothing more than
being able to advertise a larger number.

~~~
bsder
Ah. Sorry. I didn't do the math.

------
nkurz
_No one can see X-rays (or infrared, or ultraviolet, or microwaves). It doesn
't matter how much a person believes he can. Retinas simply don't have the
sensory hardware._

The author seems to have stumbled into a poor example, as a recent study shows
that humans can indeed see infrared light using an unexpected process. Should
we read anything into the audio case from this? Probably not, but it's a sign
that even those who are sure they are right because they have science on their
side should retain some degree of openmindedness.

    
    
      Human infrared vision is triggered by two-photon chromophore isomerization
    
      This study resolves a long-standing question about the   
      ability of humans to perceive near infrared radiation (IR) 
      and identifies a mechanism driving human IR vision. A few 
      previous reports and our expanded psychophysical studies 
      here reveal that humans can detect IR at wavelengths longer 
      than 1,000 nm and perceive it as visible light, a finding 
      that has not received a satisfactory physical explanation. 
      We show that IR light activates photoreceptors through a 
      nonlinear optical process. IR light also caused 
      photoisomerization of purified pigments and a model 
      chromophore compound. These observations are consistent  
      with our quantum mechanical model for the energetics of 
      two-photon activation of rhodopsin. Thus, humans can 
      perceive IR light via two-photon isomerization of visual 
      pigment chromophores.
    

[http://www.pnas.org/content/early/2014/11/25/1410162111](http://www.pnas.org/content/early/2014/11/25/1410162111)

~~~
sillysaurus3
Openmindedness isn't the same thing as being open to accepting a mistaken
belief. If no one can tell the difference between 24-bit and 16-bit, then
24-bit simply doesn't matter. The only way to know is with a controlled
experiment that checks whether the people can detect no better than chance
whether music is 24-bit.

~~~
nkurz
_Openmindedness isn 't the same thing as being open to accepting a mistaken
belief._

Perhaps not, but I think it's very close. I'm not advocating for noncritical
acceptance of anything, but making the point that any jump from "X is a
physical law" to "Y is impossible" depends critically on the assumption that
the only way to achieve Y is through X, and that there is no possibility of a
back door that entirely sidesteps X. This not be openmindedness exactly, but
failing to account for this possibility strikes me as an example of
closedmindedness.

 _The only way to know is with a controlled experiment that checks whether the
people can detect no better than chance whether music is 24-bit._

Which people, which music, under what circumstances? And whose results do you
trust? I feel safe guessing that many companies selling high-end audiophile
quackery claim to have done tests showing that their equipment makes a
positive difference to sound quality. Some of them are simply lying, some are
misinterpreting their data, others have something real too small to be
reliably measurable, and a tiny remainder might have a genuine breakthrough
because they are approaching an unsolvable problem in a way that sidesteps the
previous barriers. The question is how much openmindedness is right to account
for this probability without being overwhelmed by the garbage.

 _If no one can tell the difference between 24-bit and 16-bit, then 24-bit
simply doesn 't matter._

I'd probably make the bet that there exist certain 24-bit sound files that
certain listeners can discern from the same sound file that has been
downsampled to 16-bit. While the difference may well be too small to be
actually considered "better", I don't think that any physical laws that
prevent this. I think it would be fun to see such a test. This Youtube video
on "overtone singing" might offer insight on the sorts of effects that might
be enhanced by the greater bit depth:
[https://www.youtube.com/watch?v=UHTF1-IhuC0](https://www.youtube.com/watch?v=UHTF1-IhuC0)

~~~
aidenn0
> I'd probably make the bet that there exist certain 24-bit sound files that
> certain listeners can discern from the same sound file that has been
> downsampled to 16-bit.

I would take you up on that bet. This has been tried before, and no difference
was found, even when dithering wasn't used! The noise floor on 16 bit audio is
around -96dB. There are very few HiFi systems that can manage that. Even in
the highly unlikely event that there are listeners that can distinguish it,
it's likely any difference will end up eliminated by noise in the analog
components.

~~~
nkurz
I probably should have written 16/44.1 vs 24/192, since I was thinking mostly
of the waveform 'beat' interactions as shown in the linked video. Do you feel
those are also indistinguishable? I can't afford an actual bet on it, but I'm
interested enough to explore a bit and see what I can find.

~~~
harshreality
Unless your listener can hear frequencies above 22.05kHz, it's theoretically
impossible because the sampling theorem says 44.1kHz sampling can perfectly
reproduce all frequencies below 22.05kHz.

Any differences within normal human audible frequency ranges must be caused by
imperfect DAC. Agreed?

(No, I don't have a perfect DAC, but if the audible artifacts are because of a
DAC that produces less perfect analog waveforms below 22kHz when fed a 44kHz
source rather than a 192kHz source, isn't that squarely the DAC's fault? It
should also be made abundantly clear that this is a hypothetical. Is this
actually a problem? Has anyone simulated an analog waveform from 44.1kHz
sample, compared it to oscilloscope readings from a decent quality DAC, and
noticed theoretically audible differences?)

~~~
TheOtherHobbes
Do you have a perfect DAC we can use as a reference?

------
corysama
Anyone here who has not already seen Xiph's Digital Show and Tell (
[http://xiph.org/video/vid2.shtml](http://xiph.org/video/vid2.shtml) ) should
do themselves a favor and sit down for a watch. It makes sense of a lot of
mysteries and misconceptions around digital audio.

~~~
masklinn
Starting with the first video of the series is probably a good idea, it's
aptly named "a digital media primer for geeks"

[http://xiph.org/video/](http://xiph.org/video/) for the series.

~~~
hackcasual
Though vid2 was specifically made to address misunderstandings from this
article.

------
lmm
You can't simultaneously say that you're dithering to represent low amplitudes
while also saying you're keeping enough samples to capture all audible
frequencies. Dithering doesn't create resolution out of nowhere, it sacrifices
temporal resolution for amplitude resolution. It's also bad for compression
(hence why modern video encodes are done at 10-bit even for output to 8-bit
devices), and worse if you want to use a source as the basis for further work
(i.e. a remix). If you want to store your signal in a simple, convenient way,
and not have to carefully tweak the levels for each individual recording, 16
bits isn't quite enough. And as the article admits, extra resolution certainly
can't hurt; worst case is the extra bits are thrown away.

Also 44.1KHz is a pain to do in realtime (there's not enough headroom to
really filter out the higher frequencies without damaging the 20KHz response),
meaning you need a separate mastering step which is inconvenient and frankly
unnecessary. 48KHz is a much more sensible standard to work with.

192KHz may be dumb, but 48KHz/24-bit is perfectly sensible. It gives you fewer
ways to make mistakes than CD-quality (44.1KHz/16-bit), and at some point the
extra space is worth that, particularly since it may well compress better than
a dithered 16-bit signal.

~~~
dietrichepp
> Dithering doesn't create resolution out of nowhere, it sacrifices temporal
> resolution for amplitude resolution.

Could you elaborate on that? As I understand it, dithering trades distortion
for uncorrelated noise, under the assumption that the distortion is more
objectionable than the noise. Where did you come to the conclusion that
temporal resolution is affected? The temporal resolution is instead related to
the frequency resolution.

I agree that 48kHz/24-bit is the most sensible for production, especially
since (as you said) you don't have to worry about the levels too much. But
when you master a track, you pay very close attention to the levels anyway, so
those 8 extra bits don't do you any good. I think most people can't hear past
12 or 14 bits anyway, unless the audio has a particularly wide dynamic range.

~~~
lmm
The article talks about representing signals below the notional noise floor
using dithering, which requires either temporal dithering or something morally
equivalent to it - if your ear is detecting an average of dither + waveform,
then it has to have several samples to average from.

------
TylerE
What's so funny is to see yet another occurrence of basically "because
Nyquist" yet fails to address that Nyquist only holds true over infinite time.
Over a window of any finite length perfect reproduction is NOT guaranteed.

This paper,
[http://www.academia.edu/8412078/Is_The_Nyquist_Rate_Enough](http://www.academia.edu/8412078/Is_The_Nyquist_Rate_Enough),
for one, refutes this.

More reading material:

[http://www.wescottdesign.com/articles/Sampling/sampling.pdf](http://www.wescottdesign.com/articles/Sampling/sampling.pdf)

~~~
Dylan16807
Sure, but they're including a healthy margin by not caring about frequencies
above 20KHz.

~~~
TylerE
The margin isn't exactly healthy at 44.1khz. The video guys went with 48khz
for a reason.

~~~
scarecrowbob
Yes, timing to video frames :D

~~~
rubberbandage
More specifically, timing to film frames, since 48kHz gives you 2000 samples
per film frame (at the standard 24fps film speed).

------
arh68
The author is largely right. It all comes down to the master. And the lossless
format. The numbers here don't matter, what matters is _will the masters be
better for 24 /96..24/192..32/384_? If yes, prefer the 24/96\. Don't prefer it
for numbers' sake.

Personally I can't ABX anything above 192kbps, lame vs flac. Very occasionally
a pre-echo reveals itself, but hardly ever. V2 is fine, 16/44.1 is fine, V0 &
320 are indistinguishable from flac, but buying anything but lossless is a
crap deal.

The 'volume knob' trick, once you notice it, makes it basically impossible to
objectively compare 2 headphones, or two amps. I can't match within a 1/4 dB
with my fingers.

    
    
        Humans almost universally consider louder audio to sound better,
        and .2dB is enough to establish this preference.

------
smosher_
The argument is fine for listening to a finished product but those absolutes
in the post have some hidden assumptions that don't hold in all real-world
scenarios.

That probably sounds like BS. Hear me out.

A common practice for DJs is to use special decks to DJ from iPods, or other
digital sources (eg. regular turntables with timecode vinyls operating a PC.)
DJing involves playback speed adjustment when there is a BPM difference
between the two tracks during a transition. 192kHz is overkill for basic DJing
and you'd probably be fine in the vast majority of cases with 48kHz, but if
the DJ is a turntablist (scratch etc.), you want all you can get when going
from zero to target speed—which is happening constantly. It sounds awful when
consumer-grade¹ audio is used. As for the ultrasonics, filtering is the answer
in this scenario. It may be quite a good thing for these DJs (and their fans)
that Apple is doing this. (I'm not under the impression that this is _why_
they're doing it.)

A lot of the music I listen to uses samples that are played at something like
half speed, or tuned down at any rate. I tend to tune samples down by about a
fifth myself. Point being: a lot of detail that would otherwise be present
with higher samplerate source material goes missing. It doesn't help that
tuning down like this dumps a good portion of the low end.

There are also neat ways of exploiting nonlinearities from ultrasonic sources,
which I use, but that's harder to describe.

¹not meant as a derogatory remark.

~~~
GrantS
I had a similar reaction. In addition to the article's assumption that the
consumers and producers of audio are distinct groups with no overlap, there is
the assumption that humans will be the only ones listening (rather than
computers, e.g. Shazam).

Those two factors are probably not enough to change the way ALL music is
distributed, but they deserve to be acknowledged and dismissed with proper
evidence as well.

------
JoshTriplett
Such "studio" formats don't make sense for end-user listening, but they make
sense as inputs to further processing, mixing, etc. Having a pile of headroom
in frequency and amplitude means that after further processing you can
subsequently output a sensible 16-bit 48kHz file without loss. If you _start_
with a 16-bit 48kHz file and then do a pile of processing, you won't
necessarily preserve the same degree of quality.

~~~
scw
Most of the places selling these files don't include disclaimers about how the
additional quality is only useful in studio conditions -- they use it as a
differentiator in the marketplace for end listeners. The article does mention
the need for higher quality in production environments:

> Also, there are (and always will be) reasons to use more than 16 bits in
> recording and production.

> None of that is relevant to playback; here 24 bit audio is as useless as
> 192kHz sampling. The good news is that at least 24 bit depth doesn't harm
> fidelity. It just doesn't help, and also wastes space.

------
jahnu
I can't emphasise enough how good the linked presentation video is...

[http://xiph.org/video/vid2.shtml](http://xiph.org/video/vid2.shtml)

That is how you do a presentation!

------
tunesmith
The summary here is to just rip everything losslessly and then to go ahead and
use 44.1/16 since it's actually better in some ways and not worse in others.

~~~
RyJones
Indeed. I rip everything losslessly and batch re-encode when I switch music
players (man, I miss my Zunes)

~~~
shmerl
+1. I store originals (rips and etc.) in FLAC and then encode in Opus (around
140 Kbps) to actually listen to. 140 is an approximation of transparency
level, it can be lower according to these tests: [http://listening-
test.coresv.net/results.htm](http://listening-test.coresv.net/results.htm)

~~~
rikkus
Curious: Are you playing back with Rockbox?

~~~
shmerl
Yes, I have it installed on a Sansa Fuze player. Waiting to get Jolla 2
handset when it will come out, and then will probably play it there. I'm sure
though that VLC would handle it just fine on any Android device as well.

~~~
rikkus
Great, might look into getting one of these myself.

Thanks for that!

------
mxfh
I think most effective way to improve sound quality is to get a good DAC as
close as possible to the output. Headphone amps with integrated DACs do
wonders for little money. When possible go for XLR on the last mile to the
speakers (good neutral studio speakers) to cancel out distortion from external
electromagnetic pulses. To me differential, or balanced signalling is still
the most clever, yet so simple, analog information preserving method I ever
heard of.
[http://en.wikipedia.org/wiki/Differential_signaling](http://en.wikipedia.org/wiki/Differential_signaling)

Also; of cause 24bit increases the resolution of the signals amplitude, that's
exactly what it does by definition, that it sets the noise floor is only a
function of that. To stress the visual analogy: it's like looking at 16bit
images with banding and then on 24 bit images. Of cause the effect is not very
pronounced but it's there. Especially if you have a high dynamic recording of
a soundscape but you would like to _zoom_ in into a certain volume range like
that of the human voice it's good to have that extra resolution not only in a
studio environment. Think about it like developing a picture from RAW into
JPEG to stress this analogy again, this doesn't make a difference for
compressed pop music but for uncompressed recordings of live performances with
analog instruments or natural soundscapes it does. You can then choose at
which volume range your most at home for listening and do the compression or
not.

I like to think about 24bit audio like having access to the RAW files of
images, it doesn't matter in most cases, and most likely you should leave the
mixing to professionals artists for the intended effect, but it also enables
you to experience the sound in many more different ways.

~~~
dietrichepp
> When possible go for XLR on the last mile to the speakers (good neutral
> studio speakers) to cancel out distortion from external electromagnetic
> pulses.

A couple minor corrections: XLR is just a connector, the equipment has to be
balanced and there's plenty of equipment on the market with XLR jacks but
unbalanced signals. Second, electromagnetic interference does not cause
distortion, but it does cause noise.

Balanced signals are far from a magic bullet. It is basically a tool for
solving a couple specific problems: interference and ground loops (NOT
distortion). In a typical home setup, interference and ground loops will not
be a problem, because we're just talking about plugging a CD into a stereo
that's a few inches away, and they're both plugged into the same power strip.
Balanced cables are more helpful if you have audio equipment plugged into
different mains circuits, or drawing lots of current, or transmitting signals
over long distances (more than 10 meters). For your home stereo, regular RCA
cables are sufficient.

> this doesn't make a difference for compressed pop music but for uncompressed
> recordings of live performances with analog instruments or natural
> soundscapes it does.

I'm sorry, but this is just ridiculous. The noise floor of 16-bit PCM is -96
dB. Your living room, if it is very quiet, has an ambient noise level of 30 dB
SPL. Or you could suppose that you spent money soundproofing your rooms, and
you live out in the country, and it's as quiet as a professional recording
studio, at 20 dB SPL. Now, turn up the CD player until the noise floor is
audible. What happens when you play music? It will be at 110 dB SPL, which is
the same sound level as sitting next to a chainsaw.

Now, the benefit of 24-bit audio is that you can play music louder than a
chainsaw (> 110 dB SPL) and still have parts of the music that are quieter
than a whisper (< 20 dB SPL). Even without compression, it is rare for actual
live performances to have that kind of dynamic range. Pianos, for example, are
simply not physically capable of it, with typical microphone technique.

~~~
mxfh
In a professional audio context XLR (as in RS-297-A) is pretty much equivalent
with balanced audio, of cause this requires a balanced output and signal
chain. Just as well as a stereo phono connector can be used for a balanced
mono signal.

You surely heard a GSM pulse on your speakers before; that's not just noise.
Especially when living in an apartment where you can't control what kind of
cold fusion reactor your neighbor from hell is running, this a significant
improvement over RCA and extensive shielding also in a home setting.

I'm not talking about using the full dynamic range in a linear fashion, more
about doing the compression at home and not in the studio. Like being able to
listen to a whisper, a piano and a chainsaw from the same recording, recorded
at it's original amplitude, at playback mapped to a pleasant listening range
at full fidelity, yes that's not the normal casual use case but that's what a
higher bit depth enables.

~~~
dietrichepp
In a professional context, XLR is not the same as balanced. I have several
pieces of equipment with balanced connections, but only a minority of the
balanced connections are XLR. Most are TRS.

I think you are using a different definition of "noise". "Noise" is unwanted
sound.

The GSM noise you hear is caused by demodulation of GSM radio, in the 800-900
MHz range. At these frequencies, even a very small wire works well as an
antenna. For example, an 8 cm wire makes a quarter-wave antenna. In my
experience, there is often a trace of at least 8 cm within an amplifier or
monitor which picks up the signal. In these cases, balanced connections do not
help. The nonliniarities in semiconductor components then demodulates the GSM
signal into the audio band. The worst offenders at picking up GSM interference
are cheap amplifiers and radios. Good equipment is well shielded and doesn't
suffer from radio interference, and in my experience, using unbalanced
connections over short distances (1-2 meters) won't change that.

Balanced connections fix ground loops and let you run cables over long
distances. That is all. They don't save you from GSM demodulation.

~~~
mxfh
Exactly my point in the first post in this thread, the reason why getting the
DAC close to the output is a good idea in my opinion to minimize those types
of opportunities for interference on otherwise reasonably priced equipment
(DAC/headphone amp plus decent headphones for under USD400, or DAC with
balanced outputs and active monitors for under USD1000). Balanced cabling and
signalling is only a bonus but recommended for active studio monitors.

------
Htsthbjig
I have worked a lot with audio programming and while most of the article is
right in what is known, it could be wrong at what we don't know. Technical
myopia(you see too much of what you know but you don't see the big picture).

We are using HDR in pictures even when the eye could not differentiate between
HDR colors because it adapts to the general luminosity of the image but
CAUTION the general luminosity level affects lots of biological cycles like
the circadian rhythm.

In normal pictures we discard this info, but this info is enough for a person
to differentiate a picture from a real image in the real world.

Also natural sensors work different than our eyes and ears.

This difference makes HDR a necessary because we can see a huge shadow along
with an illuminated area at the same time because our cones in the retina
adapt locally, but if you make a single picture we can only choose to picture
the bright areas, making shadows too dark, or choose the shadows making the
bright areas appear too white.

Artificial sensors linearize over a fixed level. Nature sensors are really
continuous exponential, even touch.

The same happens with our ears. So you are doing an spectral analysis using an
arithmetic frequency decomposer called the FFT?

Well, sorry to burst your bubble but the cloclea frequency analysis runs
circles around anything we have. It does a geometric analysis, and also does
it locally. Using just a single tone as an example is a fallacy. Most real
sounds are not a single tone but changes in lots of frequencies at the same
time.

The law of diminishing returns applies to sound and video, we have a good
enough experience for what we want to do, but by no means it is perfect.

Ask John Carmack that is trying to create an immersive experience. Sound is
one of the big problems. Yes, you can understand the sound, but you know that
it is not the real sound you hear in the real world.

~~~
Intermernet
Not disagreeing with you, but one of the main reasons for using HDR in film is
comparable to using 192/24 (or even 192/32) in audio. It provides greater
dynamic range for post-processing.

The final "rendered" product will usually still be at normal dynamic range for
film, and 44.1/16 (or 48/16) for audio.

------
leoc
The most interesting development in terms of surround audio may be the Oculus
Rift. A number of people were working on combining high-quality VR HMD head-
tracking with headphones and a spatial audio system
[http://www.mee.tcd.ie/thrive/](http://www.mee.tcd.ie/thrive/)
[http://www.technologyreview.com/news/527826/microsofts-3-d-a...](http://www.technologyreview.com/news/527826/microsofts-3-d-audio-
gives-virtual-objects-a-voice/) . Then Oculus licensed a spatial audio system
itself [http://www.roadtovr.com/oculus-rift-
dk2-realsense-3d-audio-p...](http://www.roadtovr.com/oculus-rift-
dk2-realsense-3d-audio-preview-download/) and announced that the first
consumer version of the Rift would have integrated headphones.

------
danbmil99
Not sure about this one. I'm pretty sure at one point I could hear the
difference between 44.1 and 48 khz sampling. I agree 192 is overkill, but 44.1
is just above the Nyquist limit. At that resolution, the top breathy harmonics
of a piccolo are only getting 2-3 samples per cycle, which seems to leave room
for some possible aliasing if you are not 100% sure about your filters. So why
not just go crazy and throw 4X samples at the problem, eliminating any
question of proper anti-aliasing?

As to # of bits, the issue there is the wide dynamic range of music, and the
fact that our ears can adjust to this wide range. Probably you would get the
same effect as 24 or 32 bits with the right dynamic range adjustments, but
then we'll have to argue about which algorithm is "right". A surfeit of bits
just makes the question go away.

------
asimpletune
So I appreciate everyone's point of view and applaud using an empirical
approach, for those of you who share the author's point of view, but I
disagree unfortunately. For those of us who have worked in DSP, either using
it or implementing new things with it, there's s highly mathematical reason to
record the source of you audio with a higher sample rate than what the author
suggests is a generous maximum.

It has to do with waveforms and how continuous they are. So, for starters,
true, if you have a perfectly continuous wave form, at 22K, then your sample
rate must be at least 44K. In fact, with sample rate of 44k you can perfectly
discretize a continuous wave form, like a sine wave.

Does you see the problem with this? Sounds are not always continuous! If you
look at the waveform of a violin, distorted guitar, cymbal, etc... They're
very jagged. To effectively approximate these analog waveforms as a finite set
of sums you need a much higher sample rate. It makes s HUGE difference, trust
me.

So basically, technically speaking, 44K works just five if you only listen to
music made by orjan pipes and penny whistles, but most sounds are very
complicated, and to be properly captured you actually need a higher sample
rate. It's simple and mathematical. Also, this whole "44.1K is all you need
and if you don't agree with me then you're dumb and don't understand math" ra
ra ra has been going on all over the Internet for ages, and while u appreciate
the motivations that people may have, it gets a little annoying. Basically,
instead of immediately jumping to the conclusion that people's ears are wrong,
maybe the more patient and mindful approach is to ask oneself, "why does my
mathematical knowledge of a subject fall short of explaining what many people
seem to experience?".

Note: everything I said was regarding the source of capturing a sound. There's
an entire science behind compression and all that sauce.

Also Stanford's DSP lectures (available online) explain this much more
indepth, albeit abstractly.

~~~
kevincennis
As a former recording engineer, I completely agree that there's value in
capturing audio at better than 16/44.1. 24-bit means I don't have to care so
much about "filling up the bit bucket" when I set input levels because I have
a lower noise floor. And if I'm doing any DSP, obviously I like having more
information rather than less.

But I'm not at all convinced that _distributing_ recordings at better than
16/44.1 has any real benefit. I've done some blind tests before and was never
able to reliably beat 50/50 on figuring out which tracks were "higher quality"
\- and while I certainly don't have the best ears on the planet, I feel pretty
confident that my hearing is more developed than the average person. Not to
mention the fact that probably 90%+ of consumers are listening on systems with
poor speakers and worse DACs.

I often hear audio people _explain_ why music should be distributed at higher
bit and sampling rates, but I have yet to see anyone who can reliably tell the
difference - especially on a consumer-grade system.

Edit: Downvotes, really? Was there something objectionable in there?

------
inetsee
I remember when CDs first came out and Neil Young was very critical of their
sound, and I believe he was entirely justified in his criticism. When they
first came out CDs were Record Company's poor stepchildren, and they were
treated very poorly. I believe the engineers mastering CDs were given tapes
that were several generations away from the original master tapes, and I
suspect they may have even been already equalized for vinyl. No wonder
audiophiles preferred the sound of vinyl over the sound of early CDs.

Nowadays CDs are made from digital recordings to digital masters to digital
discs (DDD). I love Neil Young's music, but his grasp of the details of
digital audio recording and reproduction is not particularly strong.

~~~
analog31
I suspect something else was happening too: Recording techniques were adapted
to the problem of making something sound good after recording, mastering,
cutting, and playback. This probably meant getting away with some shortcuts
that would be masked by subsequent processing. The perfection of the digital
medium unmasked those things.

------
MBCook
I wonder how many people see 24/192 and think "24-bit, 192kbps" sound instead
of "24-bit, 192khz sound", getting the units wrong.

MP3s are (often/traditionally) 128kbps, so 192kbps would be better, and 24 is
more than 16, so it _too_ must be better.

For the people who do get the units, you still have the 'more is better'
problem. We've finally gotten past the point of everyone trying to make ultra-
compact 30MP cameras because consumers have realized that their current camera
is 'good enough' and that more MP doesn't always make the picture sharper.

Could this just be the same thing in the sound world?

------
bjornsing
TL;DR, but skimming through an interesting problem comes to mind: The OP
"orthogonalizes" the question of the sample accuracy's (24 vs 16 bit) and
sample rate's (48 vs 192 kHz) impact on quality, answering one independent of
the other. But even with my limited background in mathematics it's quite
obvious that that approach is not entirely correct: the Nyqvist theorem only
really applies when you have infinite sample accuracy. It would be interesting
to see an analysis of how the two interact; i.e. how the discretization error
impacts the highest representable frequency.

~~~
derf_
The _highest_ representable frequency really is the Nyquist rate: you only
need 1 bit of sample depth to generate a digital signal whose corresponding
band-limited continuous signal is a sine wave that oscillates at that rate.

Sample depth roughly tells you how far away a sample can be before the
magnitude of the ideal sinc function used to reconstruct the continuous signal
falls below your quantization threshold. That distance in turn gives you an
idea of the frequency _resolution_ you have... You could in theory run an FFT
over that many samples and detect that the corresponding change was not just
due to quantization noise.

------
caseydurfee
Fascinating article. Is there any reason to believe that upsampling an audio
file should produce a better analog signal than playing it at its native
sample rate?

I find that playing high quality mp3's through an upsampling DSP filter and a
24/192 DAC seems to produce a better listening experience. (As the article
points out, this could be due to confirmation bias, or the filter making the
music a tiny bit louder.) Intuitively it makes sense to me that the DAC
sending signals twice as frequently to the headphones would produce a smoother
analog signal, but is that actually true?

~~~
masklinn
> Intuitively it makes sense to me that the DAC sending signals twice as
> frequently to the headphones would produce a smoother analog signal, but is
> that actually true?

No. Watch the Digital Media Primer and Digital Show & Tell, Monty (the
article's author) explains how digital sound encoding works and why the analog
signal will be reproduced beyond your ability to notice imperfections either
way: [http://xiph.org/video/](http://xiph.org/video/)

And that's for native 24/192, in your case since it's been upsampled the
24/192 signal _can 't_ have more information than the original 16/44.1, since
that's all the information that went into it.

~~~
LnxPrgr3
The engineering reason to upsample is to simplify the analog anti-aliasing
filter on the other side of the conversion by giving it a wider transition
band to work with. It also means one analog filter can handle a wide range of
samplerates.

masklinn is right though—from a consumer perspective, there's no real reason
to care how the conversion's done. Hopefully your DAC was designed by a team
that knows what they're doing and will handle whatever rate you feed it.

Of course, if you have one of those obnoxious devices that can't actually
clock below 48kHz or a mixing daemon configured to run at 48kHz, you're stuck
oversampling anyway, and depending on your environment that might be done
anywhere between amazingly well and linear interpolation. Many years ago I was
stuck playing that game on Linux—in that case, there's an audible benefit from
using a better designed upsampler in your player.

But if you're equipment's not broken, it's a waste of CPU cycles.

------
mcguire
Completely unrelated question:

" _Can you see the Apple Remote 's LED flash when you press a button [4]? No?
[Some other remotes] may be just barely visible in complete blackness with
dark-adjusted eyes [5]. All would be blindingly, painfully bright if they were
well inside the visible spectrum._"

Ok, what happens if you point a remote at an infrared camera?

A little research on the FLIR website shows cameras with a range of
7500-13500nm, which would be 22-40THz (?); the remotes are 300-380THz. Sigh.

------
mxfh
This is just so anthropocentric, of cause the true selfless audiophile wants
his music to be experienced in it's full natural range by bats and dogs just
as well.

------
rspeer
Monty's example files made me realize that my computer's sound chip is
probably kind of lo-fi.

No, I couldn't hear anything in my headphones when I played the 30Khz/33Khz
tones. In comparison, I heard something after it _stopped_. Somehow it was
quieter during the playback.

Does it make any sense that my computer is better at making no audible sound
when it's asked to play a loud inaudible sound, than when it's being asked to
play nothing?

~~~
TD-Linux
It is likely a power saving measure. There is a slightly audible click when
mine turns on and off automatically, and a bit of noise when it's off.

------
shmerl
Some related comments:
[https://xiphmont.livejournal.com/63490.html](https://xiphmont.livejournal.com/63490.html)

------
dzhiurgis
I am sorry, but the authors' examples and parallels doesn't make much sense to
me.

First of all, what does sample rate has even remotely to do with hearing
range?

If we are comparing this to visual information, perhaps it would be best to
draw parallels with video. Namely, sample rate would be equal to frames per
second and bit depth to colour depth. In both areas we are still seeing
improvements from TV manufacturers (100Hz TVs) and gamers are still racing to
get better fps so it looks more realistic.

The question then would be, why 25 fps is the minimum _average_ human can
withstand? What are the limits here and can it look more realistic for certain
people with more frames per second.

After all, this race for fidelity is just attempt to get more accurate
representation of analog sound, which opens different problem - absolute all
of todays music is electronic - produced from samples and samples above 44KHz
are simply not available. You may find some vinyl rips of more classical music
(which does explain the phenomenon why so many rock stars would start
listening to classic music - after getting rich and dumping money into
expensive sound systems they'd discover elegance beyond _distortion_ ).

Saying all this fidelity does not matter is kind of equivalent of throwing
away all the vinyls (although transistors play a role here too).

~~~
smosher_
> If we are comparing this to visual information, perhaps it would be best to
> draw parallels with video.

But it isn't. We perceive pitch more like we perceive color than how we
perceive motion. "Motion" in audio is well below the frequency of pitch. Same
goes for video really, but in video the framerate has nothing to do with the
color, unlike how the sampling rate has to do with sound frequency in (
_cough_ most) digital formats. If color in video was sampled the way pitch is
in audio, we would have framerates in the high terahertz range.

~~~
smosher_
Can't edit: on topic, accurate response -> downvotes.

Stay classy HN.

------
lucisferre
Should this have [2012] added to it?

------
vidoc
One of my favorite is the noise harvester by PS audio.

[https://www.youtube.com/watch?v=cJGUSHBuE_0](https://www.youtube.com/watch?v=cJGUSHBuE_0)

I could easily imagine the CEO of this company sell eternal life elixir in the
19th century wild wild west.

------
slowmotionsleep
Is it possible this was just a move to get advertising space in audio files,
like with devices that use ultrasonic frequencies to trigger events in
devices? (For example, [http://lisnr.com/](http://lisnr.com/))

------
mhewett
Self-deception has been going on for a long time. See the Wikipedia entry on
N-rays:
[http://en.wikipedia.org/wiki/N_ray](http://en.wikipedia.org/wiki/N_ray)

------
lfmunoz4
I know that people cannot perceive sound at above 20Khz but has this verified
with brain imaging? Just to be 100% certain that you are not subconsciously
hearing something above 20Khz.

------
mark-r
The entire argument is based on the premise that the ear can't detect content
above the limit of 20khz. While it is no doubt that few if any ears can hear
independent tones above 20khz, what about detecting the _shape_ of tones below
that point? I don't think anybody's proven that the overtones of notes below
20khz are unimportant, or that the ear doesn't use edge detection for example
to determine phase differences for location determination. These would require
faithful reproduction beyond 20khz.

Also to clarify a technical point, sampling at 192khz doesn't extend the
frequency response to 192khz, it only extends it to 96khz (Nyquist in action).

------
ecocentrik
If you're listening to music and it still doesn't sound like the musician is
sitting right next to you in your living room, then there's still room for
improvement.

------
alkonaut
24bit absolutely makes sense in the age of streaming devices pushing sound to
digital inputs on powered speakers. Simply for convenience reasons I want to
adjust the playback volume already at my phone. I have no analog volume
control (or rather, it's a configuration knob you set up once)

If you play back a sample at 1/8 the volume, you still have 21bits of range
from a 24bit sample. Playing a 16bit sample at 1/8 volume is basically a 13bit
range, which is bad.

~~~
astrange
You might want a high bit depth audio stream if you were seriously boosting
the volume, but not if you were lowering it. At most you might want a 24bit
DAC.

And if you can't pass an ABX test, then you don't really want it. Can you?

~~~
alkonaut
Why would s higher range not be good for lowering the amplitude? If you divide
the samples in a 16bit stream by e.g 16 that means losing several bits of
range. Surely if a 12bit dynamic range was perceived as being just as good as
16bits then we'd have that to begin with?

To be clear, when I say "lower the volume" I mean lowering the amplitude of
the digital samples, passing them through a high quality dac and amplifying.
I.e sliding the volume slider down on a iPhone using AirPlay to a
Toslink->amp.

------
upofadown
The upside here is that perceptional encoders could do a really super job of
compressing such material... :)

------
donpark
Sell to people what they want, not what you think they need. It's all about
perception not whether there are basis behind the perception. Even Placebo
effect needs equivalent of little pills as props. Higher the price, more
they'll get out of the effect. If some nut can hear God's voice from a lump of
rock costing $1M, who are we to take away his 'talent'?

------
bjt2n3904
While the 30/33 kHz inter modulation test is interesting, how would the 30/33
kHz tones get there in the first place? Aren't there low pass filters and
shielding on most studio equipment to attenuate EMI? Not to mention, the inter
modulation products are 60 dB below the 30/33 kHz tones, I'm sure you wouldn't
be able to hear them...

~~~
CUViper
I think the point is that some people assert that they _want_ the ultrasonics,
as if they could tell the difference, so you wouldn't filter it. Then the
presence of those ultrasonic tones creates the modulation in a range that you
can hear. Likewise, the 60dB difference doesn't matter because you can't hear
the signal at all, so you're going to notice the noise.

------
rplst8
There are a couple of things that I would like to point out about digital
sampling.

Early on in digital audio, anti-aliasing filters were awful. The ADCs weren't
very good either, but the AA filters were bad. Using a higher sampling
frequency was one method to help this problem by providing more padding in the
frequency domain so that a more gradual AA filter could be used.

Later, this was completely OBE with the advent of oversampling. Most ADCs in
use now are oversampling delta-sigma converters operating a very high sampling
rates that perform decimation on the output to provide a 16 or 24 bit waveform
at the normal 44.1kHz or 48kHz sampling rate. Delta-sigma converters, and high
sampling frequencies are actually the basis for Sony's direct stream digital
(AKA Super Audio CD).

Today, you can be reasonably assured that an ADC will provide a very clean,
low noise, noise output in the vast majority of cases. For music playback,
this never mattered anyway. Whether or not the transport of digital audio is
at 44.1 kHz or 1 MHz, the quality to the human ear will be the same as long as
it was sampled correctly and accurately.

That said - sampling at 44.1 kHz, 48 kHz, or even oversampling, may not
capture all that sound has to offer our ears. One thing that can happen with
music for instance is beating. Beating occurs when two notes are sounded at
slightly different frequencies (say 20,000 Hz and 20,100 Hz) there will be an
audible beat at their difference. Where this can come into play is in close
mic recordings. If I record guitar A) with an overtone at 30,000 Hz and guitar
B) with an overtone at 30,100 Hz, these two would have an audible 100 Hz beat.
However, if we filtered and sampled at 48,000 Hz or over-sampled then
filtered, we would lose both overtones and the ability to hear that beating.

How important is that beating? Good question - but with live music it's not a
problem and with close mic recordings it is. For recording, there are reasons
to use higher sampling rates for at least the mixdown process. I still think
it's silly to have 192 kHz audio for listening to recordings at home.

Bit depth on the other hand is something I think we could use more of -
especially with classical recordings. Audio quantized to 16 bits only provides
about 96 dB of SNR. I would much prefer having 20 or 24 bit audio to fully
encompass the actual range of human hearing that is closer to 120 dB.

The other thing that cannot go without mentioning, MP3s, AAC, and other lossy
audio formats are pretty good - but they DO NOT compare to lossless audio.
Having Google, Apple, and Amazon all step up to 44.1 kHz 16-bit audio sourced
from equal or greater source material would be huge improvement over MP3s and
AAC.

------
aaronrenoir
Inaudible sounds effect audible sounds and brain activity. I have never been
given goose bumps listening to iTunes.

[http://m.jn.physiology.org/content/83/6/3548.full](http://m.jn.physiology.org/content/83/6/3548.full)

------
shmerl
Here is another good article about different audiophoolery and scams:
[http://www.skeptic.com/eskeptic/10-01-06/#feature](http://www.skeptic.com/eskeptic/10-01-06/#feature)

------
pje
previously submitted here:
[https://news.ycombinator.com/item?id=6135839](https://news.ycombinator.com/item?id=6135839)

~~~
shmerl
Hm, 2 points and no comments?

------
andy_ppp
So if this is true and CDs contain already 'the best possible' audio that the
human ear can hear, why does vinyl sound better? Are we saying records have a
different pressing/recording or something else?

~~~
normloman
Here's why you might perceive a record to sound better than a cd:

1\. Vinyl introduces harmonic distortion into the audio signal. From a
scientific standpoint, this means vinyl has worse fidelity than digital audio.
But some people like the distortion, as it makes music sound "warmer."

2\. In the mid 90s, audio engineers began mastering music louder. In order to
maximize the perceived loudness, they use an effect called dynamic
compression, which reduces the dynamic range of the recording (the difference
between louds and softs). This trend was not present during the heyday of
vinyl. As a result, many CD versions of old records sound worse than the
original vinyl. But this has nothing to do with audio formats--just poor
mastering.

------
sramsay
Okay, so just to bottom line this for someone with only rudimentary knowledge
of audio and for whom all of this is a revelation:

I am an idiot for paying for Pandora, yes?

------
frontsideair
I loved watching their Show and Tell video. I'd like an update everytime they
uploaded a video, I wish they had a subscription button or something.

------
scott_karana
Why the title change? It is: "24/192 Music Downloads ...and why they make no
sense"

~~~
Nekit1234007
Read the <title>

------
treerock
Aren't there benefits other than listening quality to having the music
available in this format? Imagine 200 years from now, finding an old, lost
stash of music in the attic. What would you prefer, a bunch of CDs or some
drives with 24/192 FLACs on them?

------
the_solution
Disclaimer: I am NOT saying that I think 24/192 is a good idea or makes any
difference.

That being said, the author makes the error to assume that we experience music
only with our ears. But your body can _feel_ e.g. the sound of a drum.

I call for a double bling test with loud enough speakers capable of
reproducing the spectrum in question.

I highly doubt there will be an effect from 24/192 but then we will know for
sure.

------
Cthulhu_
They're silly for listening maybe, but for editing (or so I gathered) they're
quite valuable because the extra bits give editors a lot more room to play
with the sounds and frequencies and such.

It's not always about the audiophiles, you know.

~~~
corford
Which the author mentions repeatedly in his post.

------
samatman
I think this underestimates how many people are DJing. You don't have to be a
professional to want to warp a couple tracks together, and this is a case
where more samples help. Many of the 'only professionals need this' arguments
fall apart; DJing is a popular hobby.

------
cjensen
The author confuses sampling theory with reality.

Sampling theory says you can reconstruct a signal by sampling it at least
twice per cycle. So 44.1KHz is an adequate sampling rate for an 22.05Khz
signal.

Unfortunately to make Digital-to-Analog conversion work according to theory,
you must first construct an ideal analog filter which filters out everything
above 22.05Khz while leaving everything less than 22.05Khz unmolested. That's
not possible in reality. If 20KHz is the goal, you have a measly 2.05KHz to
make the filter ramp from kill-nothing to kill-everything. I'd imagine real-
world CD players with cheap filters probably kill everything above 15Khz.

In reality you want a _lot_ of headroom between half the sampling frequency
and the actual max frequency you want to pass unmolested. Even 48KHz only
grants you a 4Khz band in which to let the filter roll off.

Second, significant playback timing jitter can render the LSBs useless. At
44.1KHz with 16bit sampling, the max difference between a pair of samples of a
22.05KHz max-amplitude input is very roughly 2^15. What does that mean? If
your jitter is more than 692ns [1] you have just lost an LSB.

Sure, 24/192 is serious and unnecessary overkill. The advantage is that it has
lots of headroom. The disadvantage is that it takes more space. If you were
designing a new format today in our era of large hard drives, why wouldn't you
waste a bit of space?

This isn't gold-plated monstrous-cable properly-broken-in HDMI snake oil. The
current format isn't perfect; an upgrade is a reasonable idea.

[1] 44.1KHz period is 22.6ms; one part in 2^15 of that is 692ns.

~~~
jongraehl
Author is an expert, and correct. In the linked video he mentions that the
difficulty of creating a sharp analog high-pass filter is in practice
completely mitigated by oversampling, which is described in the Wikipedia
article on DACs.

Suppose you have a 96khz DAC coming from your computer. Surely you see that a
computer can solve for the nyquist reconstruction of some lower sample-rate
recording (e.g. CD audio) to that sample rate, and then (still in digital
96khz) remove ALL the noise between say 21khz and 96khz, at which point a
final analog filter will have an easy time leaving 0-21khz unmolested while
killing all higher frequencies, right? It's practically implied by what you
wrote. Per what OP says, that's more or less the effect of any DAC you'll use
today.

------
TarpitCarnivore
I would imagine in most cases the quality of the rip wont matter if the
production on the album, or song, isn't very good. A poorly produced and
mastered album wont sound good whether it's 128kbps or a full lossless rip.

I think similar to a lot of things there are some people out there who can
notice the difference, but the percentage is likely very small. Most probably
are telling themselves it sounds better, but as the article points out they're
likely guessing when it comes to doing A/B tests.

~~~
masklinn
> I think similar to a lot of things there are some people out there who can
> notice the difference

Have you considered reading the article? Because it's reasonably clear that
you can't "notice the difference" and _be a human_.

------
KeytarHero
24/192 playback isn't _entirely_ silly. It's probably true that you can't hear
a difference between a perfectly reconstructed 16/44.1k audio and perfectly
reconstructed 24/192k audio. But the quality of your DAC certainly does
matter, as well as the analog hardware that's after it. If a device has a
24/192k DAC, it means the manufacturer didn't just use the cheapest DAC they
could find, and it's more likely to be high-quality (plus, it shows they care
about audio quality, so they may have gone for a higher-quality analog
hardware too).

In a perfect world, cell phone and music player manufacturers would advertise
"this phone has a Wolfson DAC and uses OPA2134 opamps running at 9V in the
output stage". But the mass consumer market doesn't care about those things,
so for now all we have to go off is "this one supports 24/192 so it's probably
got better hardware" (or the alternative, "this one says it has Beats Audio so
it's probably got hyped bass and is terribly inaccurate").

~~~
derf_
Did you read the article?

"The ultrasonics are a liability during playback.... Neither audio transducers
nor power amplifiers are free of distortion, and distortion tends to increase
rapidly at the lowest and highest frequencies. ...any nonlinearity will shift
some of the ultrasonic content down into the audible range as an uncontrolled
spray of intermodulation distortion products covering the entire audible
spectrum. Nonlinearity in a power amplifier will produce the same effect."

I would not expect a willingness to exploit consumers' magical thinking to be
a good signal for quality engineering.

~~~
KeytarHero
That's what audio mastering is for. A good mastering engineer tests their
masters on iPhone headphones as well as studio monitors.

> I would not expect a willingness to exploit consumers' magical thinking to
> be a good signal for quality engineering.

Companies do this all the time, not just in audio, and they have for years. I
don't know why most consumers would need a phone with a camera more than 8 MP,
when most users will only ever display it on a 1080p (~2 MP) screen. I don't
know why anyone needs a screen with more than 300 PPI. I don't know why anyone
needs a TV with higher than 120 Hz refresh. But guess what, if Nokia puts a
41-megapixel sensor on their phone, I'm willing to bet they've also got a darn
good lens. If Google wants to put an almost 500 PPI screen on their phone, I'm
guessing they've chosen a screen that also has pretty good contrast & color.

Sample rate/depth is one thing device manufacturers can do to easily send the
message "we care about audio quality" to the general consumer market, just
like how a 41 MP phone camera tells you they are serious abut the quality of
their camera.

Obviously you shouldn't judge a phone's camera or screen by the number of
pixels alone. Unfortunately, it's much easier to directly compare screens and
cameras than it is phone DACs. I wish there was a good benchmark system for
audio hardware, but it's really hard to find accurate, unbiased, quantitative
information.

~~~
dragonwriter
> I don't know why most consumers would need a phone with a camera more than 8
> MP, when most users will only ever display it on a 1080p (~2 MP) screen.

(1) Many computers, TVs, tablets, smartphones, and laptops now have screens
with greater than 1080p resolution ("Quad HD" 2560x1440 is particularly
common), so I don't thinks it true that most will only ever display pictures
on a 1080p screen, heck, many of them will be taking pictures on devices with
greater than a 1080p screen.

(2) Often pictures, after being taken, will be cropped; so the image that will
be viewed on a screen (of whatever size) will be some subset of the full
picture taken.

~~~
KeytarHero
(1) That's beside the point. Even if a phone has a 4k display, they still
don't need a camera of more than about 8MP to display it on screen.

(2) Yeah, I get this use case, and I understand why there are DSLRs that big.
But how many cameraphone users are actually cropping their images so extremely
that they need 41 megapixels? Phone makers don't put 41 MP sensors for the
niche market of users who need cameras that good but don't have a DSLR; they
do it because the majority of their customer base thinks "the more pixels the
better".

~~~
dragonwriter
> But how many cameraphone users are actually cropping their images so
> extremely that they need 41 megapixels? Phone makers don't put 41 MP sensors
> for the niche market of users who need cameras that good but don't have a
> DSLR; they do it because the majority of their customer base thinks "the
> more pixels the better".

Actually, the Nokia 41MP sensor is sold as enabling high-power digital zoom,
which is the feature (with the associated benefit of taking clear pictures
from much further away than with other phones) of the phone most heavily
touted in the TV ads for the phones with the sensor. And digital zoom is
_exactly_ the same thing as cropping.

So, no, I don't think the actual marketing of the phone supports the idea that
41MP sensor is targeted at people using MP as a quality metric disconnected
from any concrete utility, its targeted at selling a very specific benefit.

------
whiddershins
Edit: I forgot to mention I don't think the author does a good job of
addressing how jitter can reduce the effective sample rate. This is a real
issue, and why dsd was conceived of.
[http://en.m.wikipedia.org/wiki/Direct_Stream_Digital](http://en.m.wikipedia.org/wiki/Direct_Stream_Digital)

Regardless, for non pop music, a strong case can be made for 24 bits. The
assumption that music doesn't _need_ that kind of dynamic range is based
entirely on a subset of music that happens to be what most people listen to:
highly compressed popular (rock/hiphop/edm/etc) music.

The author falls for the "this makes sense in terms of physics and biology
therefore must be true" way of thinking.

------
bsaul
I'm clearly not qualified in biology and DSP to argue with the 24/192 argument
(although i am one of those guys that will swear that it definitely sounds
better, having tried one), but here's another thing to consider :

\- Since the 2000's the general trend with music has been to compress audio as
much as possible to send it through the pipe and store it everywhere, assuming
availability was a superior concern than quality.

Therefore, the audio industry in general has been totally collapsing, in favor
of the network provider business and now cloud storage one.

\- We audiophile, have been thus pushed during the last decade to listen to
shite mp3 on our phones, to get poor audio resolution on our cable TV, and
often cry at those "boom bass" headphones and speakers advertised as being the
bests.

In this context, this new trend to high fidelity audio for everyone is simply
a miracle. Maybe we will finally see lossless audio everywhere. Maybe we will
get 4 or 5 channels audio file formats (i have fond memories of a choir song
demo included with my soundblaster audigy which put you in the middle of the
choir). Maybe all of this just means the world is ready to jump to high-res
audio hardware _before_ 4K video becomes mainstream...

------
z3ugma
I'm not familiar enough with the author's work to know if his credentials
merit it, but to me his tone is unbelievably snobbish and offputting.

~~~
sp332
He founded xiph.org, created the Ogg container, and helped write Vorbis and
Daala. His creds are pretty much the best.

Edit: For a less-snobbish presentation, try watching his Digital Show & Tell
video [https://www.xiph.org/video/](https://www.xiph.org/video/)

~~~
TheOtherHobbes
I wouldn't say being a data compression geek makes anyone a master of DSP or
of psychoacoustics.

The basic issue with 16-bit is that low level details like reverb tails and
hall ambiences or very quiet musical passages get the equivalent of 14-bit (f)
12-bit (mf) to 8-bit (ppp) sampling.

This sounds noticeably grainy and digital.

It's not about total dynamic range or sine waves, it's about the fact that
human ears can do _really neat_ source separation tricks. We can hear quiet
elements in a mix without too much difficulty.

If those elements are sampled at less than 16 bits - which they will be, if
the maximum resolution is only 16-bits - we can hear that too.

So 24-bits gives you effortlessly smooth sound for quiet passages and quiet
details. 16-bits doesn't. (Dither helps a lot, but it only takes you so far.)

Why are there still people who pretend this isn't relevant? It's not a
difficult point to understand, and it shouldn't be controversial.

Edit - the technical misunderstanding is a lack of appreciation of the
different properties of the absolute theoretical noise floor of a converter,
and the fact that quantisation noise isn't like analog noise. It's actually
more of a hyper-objectionable and nasty sort-of-nonharmonic distortion.

So as the bit resolution goes down, the sound doesn't just get buried in noise
_it also gets more and more obviously distorted._

~~~
mmastrac
If this is so relevant and obvious, why not put up two files: one 192/24 and
one 48/16 and allow people to run their own double-blind test as he notes in
the article? If you could produce a repeatable test where some number of
people can tell that one is better, that would be a powerful argument.

He's argued that people have done this test over and over, and nobody can ever
tell the difference.

~~~
TheOtherHobbes
Firstly people haven't 'done this test over and over.'

There's been exactly one serious sort-of peer-reviewed paper in the AES
journal, and that paper compared high-res _commercially mastered_ audio
sources of possibly questionable parentage with a 44.1/16 downconversion.

It also included SACD, which isn't a fixed bit depth linear PCM technology,
and has been justifiably criticised for it.

I'm not aware of any tests that compare raw high-res unprocessed recordings
with downsampled content.

Secondly, a fair comparison would be 48/16 and 48/24.

Personally I'm not very sold on high sample rates. I know there are technical
reasons why it's easier to make antialiasing filters sound transparent at 96k
than it is at 44.1k, and in practice it's not easy to pull apart practical
design from theoretical limits. (Nyquist is only ever an ideal. No hardware is
ever Nyquist-perfect.)

Basically psychoacoustics is hard. Ears are ridiculously sensitive, brains are
occasionally delusional, and marketing people lurk everywhere.

It's extremely difficult to pull apart fact from reality.

But that's no excuse for having a misleadingly superficial understanding of
the theory - which the original article does.

------
jimon
Ok, I have a 1k+$ of sound equipment, I listen to lossless, and my equipment
is 24/192 DAC in hardware (so even if I pass 16/44.1, it will get up-
converted). Now give me 24/192 music and don't tell me "you don't need it",
because I need it.

But for my mobile phone and 29$ earpods, 96 kbps 16/44.1 mp3 from soundcloud
is good enough.

~~~
reportingsjr
Uhhh, 16/44.1 can't be "upconverted" to 24/192\. Once you lose that
information is gone it is gone forever.

~~~
TylerE
Dure, but there's no reason to expect you lose anything, either, barring
implementation errors. Lots of the TV you watch looks fine, despite being a
720p signal displayed on a 1080p display. Same concept.

~~~
masklinn
He means the "information loss" from encoding the analog signal to 16/44.1

