
24/192 music downloads make no sense - LERobot
http://people.xiph.org/~xiphmont/demo/neil-young.html
======
hatsunearu
The single worst thing that harms audio quality is excessive compression. It's
ruining everything, I'd say. Heck, I'm sure everyone who knows their worth
would agree that compression harms audio quality.

Recommended reading:
[https://en.wikipedia.org/wiki/Loudness_war](https://en.wikipedia.org/wiki/Loudness_war)

edit: compression as in dynamic range compression, not data compression like
mp3 in audio

~~~
hammock
Dynamic range compression is an important part of the aesthetic of pop music
today, like it or not. Pop hits don't sound the same without it. For this
reason (unaffected by normalization on itunes/youtube) it will be slow to go
away.

~~~
Eiriksmal
Did some brief Googling to see if I could participate in a blind listening
test and hear the difference between 24- and 16-bit recordings. I found this
instead [0], an even more interesting gap to see if you can hear the
difference between 8- and 16-bit!

The source song, PSY's Gagnam Style, is the epitome of modern pop. I got a
3/10 on the listening test on a decent pair of Sennheiser headphones in a
quiet room.

Some people are commenting that modern pop pairs well with 16-bit because of
the heavy-handed mastering techniques and that older music thrives under
24-bits. Well, Audio Check offers the same 16 vs 8 test, using a Neil Young
track from 1989... I couldn't fool myself into hearing any differences between
the source WAVs at all and didn't even attempt to score the 10 soundbites.

[0]
[http://www.audiocheck.net/blindtests_16vs8bit.php](http://www.audiocheck.net/blindtests_16vs8bit.php)

~~~
semi-extrinsic
With these online tests, can anyone shed some light on how browser/plugin
codecs come into play? I recently took Tidal's test to see if I could
distinguish their lossless Flac option from 320 kbps AAC. Using a decent DAC
and some Sennheiser HD202's I managed to score 0/10, which is somehow
impressive.

~~~
rikkus
I believe in various tests no-one has been able to reliably discern lossless
audio from 320kbps mp3, even though mp3 performs comparatively poorly at lower
bitrates, compared to more modern lossy codecs.

------
earlz
I can't hear a difference between 96khz/44khz in it's raw form. However, I can
tell the difference from effects in audio mixing. The extra detail can really
make a difference in how well an audio effect VST works.

I have a 96khz/24bit interface that I use and ATH-M30X headphones, and I can
tell a difference between at least some 24bit FLAC files and 16bit highest-
quality-possible MP3s. I was mixing my own music and the difference was quite
obvious to me. The notable thing was that drum cymbals seemed to have a bit
less sizzle and such.

Now that being said, if I hadn't heard the song a million times in it's
lossless form from trying to mix it, I probably wouldn't have noticed, and
even then it didn't actually affect my "experience".

I'm one of those guys that downloads vinyl rips as well, but I do that mostly
just to experience the alternative mastering, not that I think it's higher
quality or anything. (though I have heard a terrible loudness-war CD master
that sounded great on vinyl with a different master)

~~~
ska
The article is pretty clear about this too - higher bitdepths and sampling
rates _can_ be quite useful in mixing and recording situations.

They're pointless for playback.

~~~
TheOtherHobbes
No they're not. And no matter how many times this gets linked to on the
Internet, it's still wrong.

The basic problem: the quieter a sound or detail gets, the fewer bits of
resolution are used to represent it.

In 16-bit recording, there simply aren't enough bits to represent very low
level details without distorting them with a subtle but audible crunchy
digital halo of quantisation noise.

In a 24-bit recording, there are.

Talking about dynamic range completely misses the point. It's the not the
absolute difference between the loudest and quietest sounds that matters -
it's the accuracy with which the quieter sounds are reproduced.

This is because in a studio, 0dB full-scale meter redline is calibrated to a
standard voltage reference, and both consumer and professional audio has
equivalent standard levels for the loudest level possible.

 _These levels don 't change for different bit depths_, and they're used on
both analog and digital equipment. (In fact they've been standard for decades
now.)

This is why using more bits does not mean you can "reproduce music with a
bigger dynamic range" \- not without turning the volume up, anyway.

What actually happens is that the maximum possible volume of a playback system
stays the same, but quieter sounds are reproduced with more or less accuracy.

In a 16-bit recording quiet sounds below around 50Db have 1-8 bits of
effective resolution, which is nowhere near enough for truly accurate
reproduction. (Try listening to an 8-bit recording to hear what this means.)

You might think it doesn't matter because they're quiet. Not so. 50dB is a
long way from being inaudible, ears can be incredibly good at spectral
estimation, and your brain parses spectral content and volume as separate
things.

There's a wide range between "loud enough to hear" and "too loud" and 24-bit
covers that whole range accurately. 16-bit is fine for louder sounds, but the
quieter details just above "loud enough to get hear" get audibly bit-crushed.

The effect isn't glaringly disturbing, and adding dither helps make it even
less obvious. But it's still there.

24-bit doesn't need tricks like dither - because it does the job properly in
the first place.

Now - whether or not commercial recordings have enough musical detail to take
full advantage of 24-bits is a different question. For various reasons -
compression, mastering, cheapness - many don't.

But if you have any kind of aural sensitivity, you really should be able to
A/B the difference between a 24-bit uncompressed orchestral recording and a
16-bit recording using an otherwise identical studio-grade
mixer/mike/recorder/speaker system without too much difficulty.

~~~
ska
If your mastering is done competently, you really aren't going to be able to
hear it in a realistic scenario. Which is why:

    
    
      "Talking about dynamic range completely misses the point."
    

isn't really sensible. It's the use of dynamic range that decides how much
useful resolution you have when quantizing a signal. This is really why higher
bit depths on _record_ and _mixing_ are useful - they let you be sloppier with
the inputs without losing much information before you've had a chance to work
with it. It still doesn't gain you anything fundamental but it does mean if
you got the levels a bit wrong you can salvage it. Higher bit rates here are
excellent.

There is nothing magic about 24 bits here. Record something with 48 bits but
set up your equipment all screwy so your only actually using the first 8
bits... and you've got an effectively 8 bit recording.

In real world applications the codec is giving you trouble with the low
amplitude stuff, not the quantizer. Not that in realistic situations your
equipment is likely to be able to generate this cleanly anyway.

    
    
       "24-bit doesn't need tricks like dither - because it does the job properly in the first place."
    

No. Dither isn't a trick, it is a fundamental approach to quantization error
at any depth.

On playback, the issue goes the other way around. If you've mastered things
correctly you'll be using the available dynamic range of the output in such a
way that the information content of your signal is well represented. This is
sufficient at CD rates for all practical listening scenarios.

~~~
beat
Mastering, especially modern mastering, compresses the bejesus out of the end
product. Trust me, you _do not want to listen_ to uncompressed recordings
under real-world conditions. Details will be so quiet you can't hear them.
It'll sound thin and dull. Most modern pop has maybe 5-6db of dynamic range.
Really loose, open mastering will be 20db or so.

As someone who both records/mixes albums and a live-instrument musician, a
live instrument in the room sounds utterly different than any recording. Not
necessarily _worse_ , just different. The pursuit of "accuracy" in audio
playback is childish and naive. The sound of a recording is a function of
technical limitations, compromises, and aesthetic decisions as much as it is a
product of the raw source sounds. Don't make it sound accurate, make it sound
GOOD! And that usually means a lot of compression, and often deliberate
distortion.

------
some-guy
This article hits close to home: before I became a programmer I worked as an
audio engineer at a fledgling studio in my hometown.

The amount of misinformation / junk-science in the audio world is
preposterous. There's a religious-cult of an industry that feeds off the
ignorance and placebos of its participants. I have many friends who swear by
their What.cd 24/192 FLAC vinyl rips and spend hundreds of dollars on
audiophile AC wall outlets. Not to say that there are no differences in high-
end audio equipment, but so much of what's "good" is subjective.

~~~
lfam
In the case of sites like what.cd, I think that FLAC 16/44 rips of CDs and
vinyl are useful for creating distributed backups of our cultural corpus. But
I agree that 24/192 FLACs of vinyl are ridiculous.

~~~
some-guy
I agree, in fact I very much like the sound of vinyl, but to say it's more
"accurate" or of higher fidelity and dynamic range than 16/44 is completely
false.

------
guelo
(2012)

Previous discussion
[https://news.ycombinator.com/item?id=3668310](https://news.ycombinator.com/item?id=3668310)

------
rplst8
First, let me state that I believe that CD audio, played through a modern DAC
and quality stereo equipment is pretty much the pinnacle of home audio
listening. That is to say, I think 44.1kHz 16-bit PCM audio is plenty good and
I'm in no rush to replace my CD collection, nor do I think significant
investment in higher bandwidth audio (for playback, mixing and mastering are
another story) buys you much.

That said, there's one thing the article does not address and that is
"beating", or really inter-modulation distortion from instrumental overtones.

Instruments are not limited to 20-20kHz. They can have overtones well above
this range. Additionally, note that short pulse-width signals, i.e.
transients, like drum strikes, especially involving wooden percussion, can
have infinite bandwidth. (Not really infinite, but pulse-width is inversely
proportional to bandwidth.

In a real listening environment (i.e. live performance) these overtones have a
chance to interact with one another in the air. It is possible that these
overtones may beat with one another and cause inter-modulation products in the
audible range. For an example of this, play a 1000 Hz tone through your left
speaker, and a 1001 Hz through your right speaker. You will hear a distinct 1
Hz "beat". The audibility of these are largely dependent on listening position
and amplitude, but it is possible to occur with instruments. Since most
recordings are done using a "close mic" technique (placing the microphone very
close to the source) the interactions such as this are never recorded.

However, if full bandwidth of the producing instruments is preserved, these
interactions of the overtones can be reproduced in a playback environment
given equipment having a wide enough bandwidth and degree of quality.

~~~
cynicalkane
Nope. Intermodulation distortion for out-of-range frequencies is inaudible.
The 1hz beat you are hearing is not a 1hz sound wave, it's a 1000.5hz sound
wave becoming louder and softer once per second.

The comparison of a 1hz beat to a 1hz sound should be absurd on its face: you
need about 20-30hz to become audible, and it's a low rumble more felt than
heard. Very low frequencies sound absolutely nothing like intermodulation
beats.

~~~
thescriptkiddie
Is there a difference? Audio is one-dimensional, frequency is just the
derivative of amplitude. An arbitrarily high frequency sound wave becoming
louder and softer 440 times per second is just as much an A as a 440 Hz sound
wave at constant volume. A lot of cheap audio gear even uses a "1-bit DAC"
that is just very high frequency PWM.

~~~
hamiltonkibbe
The signal you describe in your comment is very much NOT an A. look at the
fourier transform of a 1kHz signal modulated by a 440Hz signal, and you won't
see any frequency component at 440Hz, nor any integer multiple 440Hz!

You can look at your "A440 at a constant volume" example as a 0Hz(DC) signal
getting louder and softer 440 times a second, but this is the only case in
which your example holds. Amplitude modulation creates sum and difference
frequencies, so the A that you hear is 440Hz + 0Hz. if you change that 0Hz to
1kHz, you get a signal thats the sum of a sinewave at 560Hz and a sinewave at
1.44kHz, neither of which are an A.

The distinction is that the 1Hz signal is modulating the audible signal, not
adding to it, if you look at the spectrum of the sum of those frequencies
there is no 1Hz component, whereas if you added a 1Hz signal you'd get
something completely different. And in this case the amplitude of the signal
is always changing faster than 1Hz.

Edit: another way of looking at it: You wouldn't say you can "hear DC" because
you can hear an A440 played at a perfectly constant volume.

~~~
thescriptkiddie
All of that makes sense if we consider the case of amplitude modulation, which
is multiplication. But if we are talking about the interference patterns
caused by two overlapping audio signals, that is _addition_ , is it not?

~~~
hamiltonkibbe
The two signals which are "interfering" are added together. The amplitude of
the resultant signal varies sinusoidally, as the instantaneous phase
difference between the two signals goes from 0->2pi. One way of describing the
signal would be that it is a separate tone (sitting in the middle of the two
frequencies) being amplitude modulated by a signal at the beat frequency...
which is what you hear and why you "hear" the beat frequency) I went with this
way of describing the signal because you were talking about "an arbitrarily
high frequency signal getting louder and softer 440 times per second" which is
the definition of amplitude modulation.

Counterintuitively, there is no frequency component generated at the beat
frequency when you sum a 1kHz and 1.001kHz signal, its easy to test that out
with matlab, octave, scipy/numpy/matplotlib, etc. Generate the two signals,
add them together, and look at the Fourier transform, you'll see two
components, one at 1kHz and one at 1.001kHz (assuming you take a long enough
window to have that type of resolution) and no component at the beat
frequency. A third sinewave doesn't just jump out of nowhere when you add two
separate sinewaves together.

If you take the sum of those two signals and run them through an ideal
brickwall highpass filter at 999Hz so there are no frequency components below
999Hz, you'll still "hear" the beat frequency because it isn't a separate
spectral component, its just the two signals slowly going out of phase,
cancelling eachother out, and then going back in phase and boosting the
amplitude.

------
PaulHoule
It is not just headphones that are the problem, it is the speakers.

People today are often amazed when they listen to CD or turntable content
through 70's era crossover speakers. Back in the 70's you'd have a stereo with
2 "speakers" that each had 3 subspeakers for a total of six speakers. The fad
today is to have 5.1 sound with a single driver in each satellite, also a
total of six speakers. The spatial resolution increase is good for movies,
games and TV but surround sound in music is marginal. An amazing number of old
"classic rock" recordings were done in quad and anything by Donald Fagan will
sound pretty good w/ Dolby Pro Logic, there are some more recent Bjork
recordings, but almost everything is mixed for stereo and what you loose in
frequency response is not compensated by anything, except perhaps the ability
to produce more volume with more speakers.

------
aidenn0
If you want to know more, Monty made one of the best intros to digital
sampling I've ever seen:
[https://www.xiph.org/video/vid2.shtml](https://www.xiph.org/video/vid2.shtml)

------
ChuckMcM
I have a pair of Roger Sound Labs studio monitors for my speakers at home. I
got to look at their insides when a technician was replacing a blown midrange
speaker (they have a "lifetime" warranty, however that warranty expired when
RSL did). Looking at the cross over filter network I could see a network
selecting for frequencies > 20khz and it was shunted to a resistor. I asked
about it, and the reponse was exactly like the authors, by filtering out
signals higher than the tweeter could reproduce, they improved the listening
experience.

It made sense to me, and I love how the speakers sound. Understanding is not
inserting distortion makes even more sense.

------
andy_ppp
Why do high quality DACs clearly sound better then? And they sound better with
better files. Maybe it really is all in my head but I mean listening to a
£20000 hifi the other day (vinyl) really just shocked me.

I was listening to Marvin Gaye on my friends system and I could hear that
there were several different backing singers all moving and at different
distances from the microphone.

Are there any double blind trials anywhere of Vinyl/CD/24-192khz with super
high end hifi systems? Mostly I see people suggesting that these tests are
performed from the phono output of a mac with a pair of average ear buds...

~~~
lawnchair_larry
Vinyl actually has far less fidelity. You also physically change the recording
every time you play it back. Even on the same equipment, no two plays of a
vinyl LP sound exactly the same, unlike digital.

This fact alone should cause you to question your subjective experience. You
have no idea what part of that system was contributing to what you found
pleasant. Someone who knew what they were doing could probably build a $2000
system that would blow you away just the same.

And if you were playing vinyl, there wasn't even a DAC present in the signal
chain :)

~~~
cthalupa
>Vinyl actually has far less fidelity.

Vinyl mastering is sometimes better than CD mastering though, due to the
loudness war.

I would love to sell my turntable and vinyl collection and rely purely on
digital formats. Takes up less space, technically superior format, etc.

But one thing keeps me buying vinyl:

AWFUL mastering on CDs. A significant portion of LPs are released with more
normal mastering on the vinyl, while the CD will be brickwalled all to hell.

I listen to metal, and rock as a broader genre is particularly bad about it.
One of my favorite albums of last year, Fallujah's The Flesh Prevails, had a
dynamic range of 2 to 3 on almost every track on the CD. The vinyl master? 9
to 10. Still not great, but leaps and bounds better. The CD actually clips if
you convert the songs into MP3.

Until they go back to not murdering CD mastering, I'll continue buying vinyl
:(

(I know your comment isn't directly about vinyl being bad or anything - I just
have a compulsion to bitch about the loudness war any chance I can)

~~~
Zitrax
How do the versions on streaming services compare, are they usually copies of
the CD masterings ?

~~~
cthalupa
Usually the CD master.

The optimistic people have said that iTunes and YouTube not allowing the high
volume compression will kill the loudness war [1][2], but, it is still
happening [3].

[1]
[http://www.digitalmusicnews.com/2013/10/28/itunesloudness/](http://www.digitalmusicnews.com/2013/10/28/itunesloudness/)
[2] [http://productionadvice.co.uk/youtube-
loudness/](http://productionadvice.co.uk/youtube-loudness/) [3]
[http://dr.loudness-war.info/album/list/year/desc](http://dr.loudness-
war.info/album/list/year/desc)

------
splitdisk
From my experience, what matters more than sample rate is 24 bit vs. 16 bit
sampling in the recording/production process. Using heavy compression and EQ
can mean that very quiet sounds can become louder, in this case 24 bit
recording is ideal. Sample rate wise, anything above 40khz is fine for most
ears (I've probably lost a few khz in the upper range anyways) Another note is
that most converters operate at a multiple of 48K, so it makes sense to use
48/96khz if you are recording. It all comes down to how much disk space you
have, and want to use up.

~~~
jchrisa
Have you tried listening to SACD? The high sample rate might not give you more
reproduction of audible frequencies, but the difference in arrival times it
can encode makes well recorded stereo stuff more interesting to listen to, in
my limited experience.

~~~
splitdisk
I would be very curious to listen to SACD on some good headphones in a quiet
room. Not sure if I've ever even seen a SACD player aside from maybe in the
Sony store 10 years ago. The trick would be to find something that would be
mastered for the format.

~~~
nandemo
Are you sure? Many so-called "universal" bluray players can play SACDs.

I got a Denon one. I haven't played any SACD on it yet (I got it for bluray),
though I guess I could easily find some at that video rental store (in Tokyo).

------
weinzierl

      Because digital filters have few of the practical  
      limitations of an analog filter, we can complete the 
      anti-aliasing process with greater efficiency and 
      precision digitally. The very high rate raw digital 
      signal passes through a digital anti-aliasing filter, 
      which has no trouble fitting a transition band into a 
      tight space.
    

I always thought _digital_ anti-aliasing filters were creatures from a fairy-
tale world. Much talked about but no one has ever seen one.

My understanding: If you have a an analog filter of a given steepness the only
way to further reduce aliasing effects digitally is oversampling. Or less
steep (cheaper) analog filter plus oversampling is the same as steeper (more)
expensive analog filter. People tend to say _digital_ anti-aliasing filters
when they really mean oversampling.

"24/192 music downloads make no sense" seems to be a thoroughly researched and
carefully written article. It explains oversampling very well, possible
confusion with digital filtering (anti-aliasing or not) is out of question.
But then it goes on to talk about _digital_ anti-aliasing filters, which makes
me afraid I could be wrong.

Do _digital_ anti-aliasing filters exist?

~~~
squeaky-clean
The digital anti-aliasing filter can only ever work on a digital -> digital
signal, but they're still useful in the analog->digital process.

> My understanding: If you have a an analog filter of a given steepness the
> only way to further reduce aliasing effects digitally is oversampling. Or
> less steep (cheaper) analog filter plus oversampling is the same as steeper
> (more) expensive analog filter. People tend to say digital anti-aliasing
> filters when they really mean oversampling

You're right, and it's actually both. The ADC can run at a much higher sample
rate with a cheaper analog filter, and then that digital signal is again
passed through a digital filter and downsampled.

------
acd
Try what age is your ears
[https://www.youtube.com/watch?v=VxcbppCX6Rk](https://www.youtube.com/watch?v=VxcbppCX6Rk)

Or generate a tone sweep in audacity. Generate->chirp
[http://www.audacityteam.org/](http://www.audacityteam.org/)

You loose the ability to hear high frequency sounds as you age.

Personally I can hear up to about 14kHZ

~~~
lstamour
Huh. Downloaded a tone generator, and found that while in the video I heard a
series of clicks at 16khz and beyond, I could in fact hear 16khz if I raised
and lowered the volume from nothing to loudest in the app. It sounded like a
whine and much harder to hear than I expected, distinguished most easily by
when it quickly went from present to not present and back. In fact, I kept
going up the scale doing that, and raising the volume, and found that I was
able to hear even 19 to low 20khz as a high pitched noise, very quiet even at
-6db. So ... yeah, probably does me no good considering that the loudness of
other pitches makes it near impossible for me to hear anything practical in
those frequencies. Of course, then I go to listen to music and wow, I can hear
all this detail. I think I trained my ears for it, or I'm losing it. ;-)

------
Johnny555
Wouldn't this question be answered with a large-scale double blind trial?

If more people prefer the sound at the higher bitrate and sampling rate, then
that's the better format, even if there's no technical reason why that format
is superior.

Much like how some people prefer the "warm" sound of tube amps, even if that
means more distortion.

~~~
upofadown
From the article:

>Empirical evidence from listening tests backs up the assertion that
44.1kHz/16 bit provides highest-possible fidelity playback.

You can read the article if you want to find the actual references. No one is
arguing that higher rates/bits produces any sort of distortion that anyone
would prefer.

------
z3t4
I can hear insects and buzzing electronic devices, and my partner thinks I'm
crazy some times. Thinking I might have golden ears I tested * my range and I
could hear up to 18kHz.

* [http://onlinetonegenerator.com/hearingtest.html](http://onlinetonegenerator.com/hearingtest.html)

~~~
rubberbandage
Honestly, depending on your age, that still could be “golden” — I’m 31, I’ve
taken very good care of my hearing, I’m very acutely aware of audio
subtleties, and my hearing range tops out around 16.5KHz. The so-called
standard upper limit of 20KHz really only applies to young children, which is
why CD audio being able to reproduce frequencies of 22.05KHz is already beyond
ideal, and calls for 48! no, 96!, no, 192! (or higher) is literally insane for
playback.

~~~
lstamour
Using a tone generator on my computer and a pair of headphones, I found that I
couldn't easily hear past 15-ish myself, then I started turning up the volume,
or playing with turning the volume all the way up, then all the way down.
Using that technique, I was able to distinguish noise and high pitches up to
20.2khz or so. So I think from now on, if I hear some whine, I'm going to
trust that it's there and not my imagination. Of course, it's also the
definition of going deaf, I suppose, that I have to turn up the pitches to
such a loud volume to hear them in the first place...

------
tonecluster
Some [consumer] digital low-pass filter can benefit from higher sampling
rates, leading to an overall better representation of the analog signal up to
20kHz. But there are diminishing returns as the filter "folds" the octaves
above 22kHz; A rate of 96k for certain lowpass filters is better than 48k, but
at some point there's little (if any) benefit by going to 192k or 384k. For
recording studios, go as high as you can in both bit-rate and bit-depth.
Especially when you're processing the signal "in the box". Give the software
as much data as possible to operate without introducing errors and artifacts.
There are diminishing returns there as well, but RTFM for (for example) UA
gear and software and you're good to go.

~~~
aidenn0
TFA mentions that for recording and mastering there is a use. Furthermore, the
headline implies it see the term "downloads" in the title.

~~~
tonecluster
Yep, I read that too. Even so, there are low-pass filters in some consumer
gear that benefit from, say, 96k sampling rates and result in better quality
sound. This does imply that at 44.1 or 48 they don't represent up to 20kHz
properly, of course.

~~~
fancyketchup
Lossless upsampling the 44.1 kHz recording to, say 192 kHz is trivial for the
reproduction equipment. That the LPF on the reproduction end wants the DAC to
run at greater than 44.1 kHz has no bearing on the sampling rate of the
distribution format.

------
rphlx
24/192 lossless is a digital Veblen good; some people will pay more for it
(and/or the HW to play it & store it), and almost all of them will enjoy it
more, if only because it costs more. Whether it actually sounds better is
somewhat tangential.

------
pgrote
So, what are the better settings for ripping songs?

~~~
LeoPanthera
I rip to FLAC, not because I think it sounds better, but simply because if
some newer better codec comes along in the future that will let me compress my
songs on my smartphone even smaller (Opus?) I don't want to have to get my CDs
out again. I can just transcode from the FLAC files.

------
S_A_P
I just recently purchased Izotope Ozone 7 advanced. One feature it has is
"codec preview" which lets you "solo" the codec artifacts for MP3 and AAC
format. Even at high but rates it's amazing how swishy bit reduction sounds.
It also made me realize what I was hearing with mp3s was artifacts from
compression. That said, it's not unlike tape hiss or vinyl noise. In fact I
think it can have its own charm and in some cases make the music sound more
full. It's also probably why 24/192 digital audio can sound so "cold" or
lifeless.

~~~
beat
From mastering records at home, I've found that in all but the most golden-
ears focused listening, I can't hear the difference between 192 bit mp3 and
44.1/16 cd quality. But 128 bit mp3 is audibly degraded and irritating.

That's a pretty cool feature for Ozone 7, for sure! I'm still using Ozone 5
and don't feel a need to upgrade, but that might make it...

------
idlewords
I love the idea the author mentions in passing of a dedicated speaker assembly
for ultrasonics. This seems like something that could be a huge margin
business, and the parts costs would be as low as you wanted.

~~~
xiphmont
You're late to the game. Such high margin products have existed for some time.
There are even published papers about them( _)!

_ The published papers tend to be by the same people making the supertweeters

------
Retra
They _are_ useful if you're resampling them or editing them, but I doubt
that's something consumer music services are overly concerned with.

~~~
saidajigumi
I'll note that's the entire point of Monty's (great) article, which has this
near the top:

 _Unfortunately, there is no point to distributing music in 24-bit /192kHz
format. Its playback fidelity is slightly inferior to 16/44.1 or 16/48, and it
takes up 6 times the space._

This has all been known to anyone with actual signal processing and/or audio
engineering knowledge for a long time now. As in, common knowledge to the
kinds of folks attending the AES conference at least back to ~2001 or so. The
high sample rate/bit depth stuff is useful for production process, but
irrelevant for final distribution.

~~~
thwest
There's a reasonable argument that fits within DSP theory that frequencies
sampled above audible range could have harmonics down in the audible range.

~~~
JonnieCache
This is addressed in the article. While theoretically relevant to some
recording applications, (overdubbing a string section one violin at a time,
why would you want to do that?) this kind of intermodulation distortion can
only harm the reproduction of mixed material.

------
yzhou
There's a big difference in impulse response with different sample rates, any
one can see it on a oscilloscope, I bet some one can hear the difference.

Those who don't have a oscilloscope can see the picture here:
[http://i.imgur.com/wY0wzcW.png](http://i.imgur.com/wY0wzcW.png)

~~~
nullc
What you are showing is _precisely_ the effect of low-passing, nothing more,
nothing less.

See the digital media primer 2 for more information on that:
[https://wiki.xiph.org/Videos/Digital_Show_and_Tell](https://wiki.xiph.org/Videos/Digital_Show_and_Tell)

If humans were able to hear audio above 22kHz (or what not) in any meaningful
way, we'd expect to be be able to demonstrate that effect in carefully
controlled studied and then that lack of low-passing may matter; but that
isn't what the best evidence so far shows.

~~~
yzhou
The low-passing with a brick wall filter on 44.1KHz audio can be a bad thing
sometimes, for example, pre-echo [https://en.wikipedia.org/wiki/Pre-
echo](https://en.wikipedia.org/wiki/Pre-echo) You won't hear the pre-echo on a
2.8MHz DSD audio.

------
lips
My third time reading this, and a new question popped into my head: Are there
any volume adjustments (on software or hardware) that take into account the
pain threshold curves? That is, volume adjustments that aren't flat, but that
will attenuate the frequencies that will cause discomfort at the lowest
volumes?

------
derefr
So, no point in 24/192 because it makes no difference in playback... but
having lossless downloads is important in part for enabling remix culture?
There's a bit of a double-standard here. Maybe I can't hear 24/192 audio, but
isn't it better input for sampling?

~~~
some-guy
The article is specifying 24/192 as useless for playback quality only. Halfway
down he addresses the benefits of 24/192 for the sake of mixing and mastering
different digital audio signals, but a final mix offers no benefit to the
human listener when choosing between 16bit/44Khz and 24bit/192Khz.

~~~
derefr
What I was trying to get across is: every file has two potential
purposes—listening and serving as input for sampling. So, if we care about
enabling "remix culture", wouldn't make sense to offer a "24/192 FLAC" option
for download, push DVDA over CD, etc., anyway?

I've never seen the hype _from artists_ about 24/192 as being about better
listening experience. It's about handing their consumers a better master so as
to encourage and enable more of them to be remixers.

~~~
some-guy
Yeah I think that's not obvious from the title of the article: 24/192 are
useful downloads for the sake of editing.

------
sliverstorm
To what can I attribute the consistently horrible quality of 64kHz streams ten
or fifteen years ago? Would that fall under the "bad encoder" bucket?

Edit: christ, I mixed up bitrates (e.g. 192kbps) with sampling frequency (e.g.
192kHz) again. I was referring to 64kbps streams.

~~~
CamperBob2
64 kHz isn't a standard sample rate -- you're probably thinking of the bit
rate of an MP3 or AAC file. A 64-kbit MP3 does sound pretty awful.

~~~
sliverstorm
Yup. Further confusing me was the fact that (if memory serves) Apple did offer
MP3's at 192kbps for a while, before upping to 320kbps.

Edit: apparently my memory is worse than I thought.

~~~
comex
Apple doesn't sell 320kbps anything, but 256kbps AAC, which is probably better
than 320kbps MP3.

------
StavrosK
Everything else is fine and good in the article, but I _can_ see the infrared
in the Apple remote (and all the other IR remotes I've tried). It's faint, but
plainly visible. Am I the only one?

~~~
glitch
Went to a dark room with an Apple Remote; let my eyes adjust for a little
while. Pressed it many times; I couldn't see the infrared coming from the LED
with my naked eye. (But the camera on my iPhone imaged the infrared from the
remote's LED.) I envy your biological wavelength detection.

~~~
StavrosK
Hmm, that's odd. I've noticed this with lots of remotes, I usually just look
at them to tell if the batteries are dead. I wonder why I can see it.

------
gwbas1c
This article really misses the facts of the Nyquest-Shannon theory.

In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency
content, high frequencies need to be phase-shifted.

This phase-shift is similar to how lossy codecs work.

For what it's worth: I'm a big fan of music in surround, and most of it comes
in high sampling rates. When I investigated ripping my DVD-As and Blurays, I
found that they never have music over 20khz. It's all filtered out. However,
downsampling to 44.1 or 48khz isn't "lossless" because of the phase shift
needed due to the Nyquist-Shannon theory.

I still rip my DVD-As at 48khz, though. There isn't a good lossless codec that
can preserve phase at high frequencies, yet approach the bitrate of 12/48
flac.

~~~
nullc
> In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency
> content, high frequencies need to be phase-shifted.

Your understanding of sampling theorem is incorrect. Sampling alone (not
quantization, of course) is completely lossless under the critical frequency.

We demonstrated this in a very clear way near the end, at about 21 minutes in,
on the primer two video:
[http://www.xiph.org/video/vid2.shtml](http://www.xiph.org/video/vid2.shtml)
where we show a square wave being phase shifted tiny fractions of the
intersample length.

------
JadeNB
It may be worth noting (though it doesn't change any of the science) that this
is from 2012.

------
fla
With nowdays bandwidths, why do we keep using destructive compression for
songs?

~~~
icegreentea
Lossy vs non-lossy compression is orthogonal to the sampling rate and bit-
depth (which is what this article is about). While the MP3 standard
effectively means sampling more than 48kHz is useless, there's no reason you
can't have a lossy comprssion scheme that attempts to capture higher
frequencies.

~~~
joosters
But the article makes a compelling case for why > 48kHz is completely
pointless.

------
lmm
Dithering is a horrible thing to be doing, and 44.1 is an awkward rate. So
while I agree that 192khz is dumb, 24/48 is a better standard than CD.

~~~
ska
No, dithering (properly) is usually what you should be doing when you
quantize.

See Vanderkooy and Lipshitz 1987 for why.

~~~
lmm
Paper seems to be paywalled. I can't imagine any possible purpose for
dithering before encoding that wouldn't be better served by dithering on
playback.

~~~
nullc
At 16 bits dithering is probably pointless for listening purposes.

What dithering does is it decorrelates the quantization noise with the signal.
Absent it, quantization generates harmonic spurs. In theory, on a very clean
and noiseless signal these harmonic spurs might be more audible than you'd
expect from the overall quantization level.

In practice, 16 bits is enough precision that these harmonics are inaudible
even in fairly pathlogical cases. But quantization eliminates the potential
problem by replacing the harmonic content with white noise.

Adding noise on playback just adds noise, it would not remove the harmonic
generation.

The _best_ kind of dithering scheme is a subtractive dither, where noise is
added before quanitization and then the _same_ noise is subtracted from the
dequantized signal on playback. This is best in the sense that it's the scheme
that completely eliminates the distortion with the least amount of additional
noise power. But it's not ever used for audio applications due to the
additional complexity of managing the synchronized noise on each side.

~~~
lmm
The article is saying you can use dithering to represent sounds quieter than
your 16-bit "0000000000000001". That's what I'm objecting to.

~~~
dTal
Consider the case of a 1-bit image. Let's say the "signal" is a smooth
gradient of black to white from one side of the image to the other. If you
simply quantize to the nearest value, one half of the image ends up solid
black and the other half, solid white. No amount of after-the-fact "dithering"
of this image will recover the original gradient - it is lost forever.

Now supposing we add noise to our signal before we quantize. A given pixel at
25% gray (which under the previous scheme would always end up solid black) now
has a 25% chance of ending up white. A contiguous block of such pixels will
have an average value of 25% gray, even though an individual pixel can only be
black or white. Thus, by flip-flopping between the two closest values
("dithering") in statistical proportion to the original signal, information is
preserved.

[https://en.wikipedia.org/wiki/File:1_bit.png](https://en.wikipedia.org/wiki/File:1_bit.png)

~~~
lmm
Sure, I know how it works. But it sacrifices resolution (spatial in this
example, temporal in the case of audio) and compresses poorly. Rather than
dithering, you should use a higher bit depth so that you can represent your
original gradient (with the desired smoothness) directly.

------
iamleppert
This is totally offtopic but I can't stand the "XXX considered harmful" stuff.
I had to rage-quit the article.

