
There is no point to distributing music in 24-bit/192kHz format. - nullc
http://people.xiph.org/~xiphmont/demo/neil-young.html
======
sjwright
I must say I get rather irritated when people spend time worrying about
dubious 'tweaker' methods to improve their audio, when the most under-
performing component of most people's sound equipment also has the lowest-
hanging fruit: The room itself.

When people ask me where they should spend money to improve the quality of
their hi-fi or home theater system, in nearly every case my response will be
something like "get a thicker rug" or "put something on this wall to absorb
sound reflections, even if it's just a bookshelf."

Beyond that, I'd tend to say something like "stop being so paranoid about what
you think you can't hear, and enjoy the damn music."

~~~
Anechoic
Agree one-hundred percent about the room (although the prescription isn't
always as simple as "get a thicker rug" etc).

The other issue regarding high-frequency sound reproduction is that in most
cases, the loudspeaker won't be outputting much beyond 22-25 kHz (assuming
very good quality loudspeakers, cheap consumer grade units might struggle to
hit a -6 dB point at 18 kHz) and even for the speakers that have usable output
at that range, the directivity at those frequencies will be so narrow that
your head will have to be locked in the perfect "sweet spot" to hear anything.

~~~
cop359
I don't really know much about this, but wouldn't the 22kHz sounds potentially
create beats in the lower frequencies?

~~~
nullc
Acoustic "beat tones" aren't "real" tones— you hear them because of non-
linearies in the ear-brain system, but you have to hear the initial tones
first. (Well, unless you're talking >>130dB SPL levels where the air starts
becoming non-linear, but then lower frequency recording would capture it fine)

If you could hear subharmonic beats from ultrasonics then it would be _very_
easy to demonstrate, alas.

~~~
zachrose
Curious, what does non-linear mean in this context?

~~~
mseebach
IIRC, linearity is when you put a sound wave frequency into the medium (air) a
some point, you can predict the frequency of the sound wave at some other
place using a linear function - meaning that there is no distortion. Non-
linear is when the physics of the medium starts screwing with that function.

------
cmer
There's a lot of scientific-sounded content in this, but unfortunately most of
it couldn't be further from the truth. I'm an ex-audio engineer and studied
digital and analog audio engineering; this has been debated to death over the
last 15 years.

Digitally recording a triangle is the best example of why 48kHz is very
limiting. The distinct sound of the triangle constitutes of a high fundamental
frequency, ballpark 5kHz and of many very high-pitch harmonics. Most of these
harmonics are above 20kHz. The harmonics are what makes it sound like a
triangle, not the frequencies below 20kHz. This is why the triangle is one of
the hardest instruments to digitally record. It always sounds like crap.

In theory, it's true that the human hear can't hear above ~18kHz, but it can
hear the influence of the very high pitch harmonics on a lower frequency.

EDIT: here's more data backing what I said
<http://www.cco.caltech.edu/~boyk/spectra/spectra.htm>

EDIT 2: typos, frequency mistake

~~~
leephillips
The linked article was accurate. You are confused.

"I'm an ex-audio engineer"

Hard to believe.

"The distinct sound of the triangle constitutes of a high fundamental
frequency, ballpark 10kHz"

That's a pretty high note - higher than the top key on the piano. But an
"audio engineer" would know that.

"many very high-pitch harmonics"

Since the next harmonic after the fundamental would be at 20khz, which only
young people can hear, and none of the others are audible to any human, I
don't understand what you are talking about.

"Most of these harmonics are 20kHz."

OK, you don't either.

"it can hear the influence of the very high pitch harmonics on a lower
frequency."

Sure....

~~~
leif
You clearly have little to no musical background, and think that your basic
math skills are a substitute. The overtones present in a cymbal or triangle
are not straight multiples of the fundamental, they are chaotic, and are very
important in determining the timbre. Anyone (and I mean that) can easily tell
the difference between a cymbal with and without a low-pass filter with the
threshold around 22kHz, because these "inaudible" frequencies are lost.

~~~
mortil
If anyone can hear it, then surely it must have been verified through a
double-blind test. Can you provide a citation?

~~~
leif
I don't know of any to point you to. They probably exist, but I haven't read
them. Let me know if you stir some up.

------
Derbasti
He raises a lot of valid points. However...

192 kHz is clearly overkill for listening. Not so for further editing of the
data.

Same goes for 16/24 bit, however, the difference between 16 and 24 bit is
actually audible.

44100 is not a bad sampling rate, but it necessitates very sharp aliasing
filters, which are audibly bad. A bit more headroom is well needed there.

That bit about intermodulation distortion is complete bogus. He talks about
problems when resampling high-fs audio data. However, you would never do that.
You would digitally process 192kHz all the way. Only your loudspeakers or ears
would introduce a high-pass filter, and a rather bening (flat) one at that.
There is certainly no aliasing going on there unless you resample (wrongly).
Intermodulation distortion is not the fault of the sample rate.

I mayored in hearing technology. Calling 192/24 _worse_ than 44.1/16 is total
BS. How useful it is is a different debate.

~~~
hackermom
_Same goes for 16/24 bit, however, the difference between 16 and 24 bit is
actually audible_

No, the difference is not audible at all. At 16 bits of depth on a normal low-
level audio signal (~0.3 volts), we're talking about less than 0.000005 volts
per amplitude step. This difference gets lost in the THD already at the DAC in
your audio output stage. Then it gets lost again in the amplifier. And again
in the cable to your speakers or headphones. And then it gets lost again in
the speaker elements. What survives in a normal low-level audio signal is
about 14 bits of resolution.

 _44100 is not a bad sampling rate, but it necessitates very sharp aliasing
filters, which are audibly bad. A bit more headroom is well needed there._

44.1khz IS a bad sampling rate for accurately reproducing anything except a
triangle wave or square wave above 5khz.

~~~
davesims
Heh, it's funny to see this late-nineties debate get re-hashed here. Also kind
of fun.

If it were true that there's no audible difference between 16 and 24 bit,
companies like Alesis, Otari, ProTools, etc. wouldn't have spent the last 15
years ditching 16 bit like an old pair of smelly sneakers. (better metaphors
welcome).

Seriously, anyone who has sat down in a real listening environment for 5
minutes A/Bing 16 vs 20 bit, 16 vs 24, etc. hears the difference immediately.
There's no question. This is why you can buy ADAT 16 bit 'blackfaces' for
$100, down from their original $4,000.

~~~
hackermom
Yes, and anyone who has ever sat down infront of an LCD flatscreen watching
their favorite movie on DVD/BD using gold-plated $200 HDMI cable instead of
$4.99 Walmart HDMI cable see the extra sharpness immediately. This is why non-
gold plated non-OFC HDMI cables are down to $4.99 a piece from their original
$49.99 during introduction.

~~~
davesims
That's cute. Obviously you've never recorded a rock band while riding the pre
to compensate for 16bit's terrible noise floor and horribly limited headroom.
You've never had the joy of ruining a perfectly good take because of that
wonderful sound it makes when the volume spikes into digital distortion
despite compressing the wazoo out of the input source. Glorious sound, digital
distortion. Run a dentist drill through an old Speak & Spell and you'd just
about have it.

You've never rented an expensive tube EQ during a mix to cover up 16bit's
grating harshness from 10k to 15k. Or tried like mad to make the bass drum
sound like a freaking bass drum and not a pie pan slamming against the back of
a plastic trash can. And yes, we had good mics and pres, all standard studio
stuff. Decent, not brilliant, converters, but it was the 16bit that was the
problem. Getting those 20bit XTs for the first time was like walking into the
Promised Land.

Sure, there's lots of marketing ploys out there, lots of snake oil. Moving up
from 16 bit was _not_ one of them.

~~~
lambda
Did you read the original article at all?

> Professionals use 24 bit samples in recording and production [11] for
> headroom, noise floor, and convenience reasons.

...snip...

> Modern work flows may involve literally thousands of effects and operations.
> The quantization noise and noise floor of a 16 bit sample may be
> undetectable during playback, but multiplying that noise by a few thousand
> times eventually becomes noticeable. 24 bits keeps the accumulated noise at
> a very low level. Once the music is ready to distribute, there's no reason
> to keep more than 16 bits.

The original article does say that yes, during recording and production, 24
bit audio gives you a lot more room to play with. That doesn't mean that you
can hear the difference between 16 and 24 bits for the final recording; just
that 24 bits give you more room to keep out of trouble during production.

~~~
davesims
Did you read the comment thread at all? I wasn't responding to the article, I
was responding to a comment:

...snip...

>Same goes for 16/24 bit, however, the difference between 16 and 24 bit is
actually audible

No, the difference is not audible at all.

...snip...

------
Anechoic
For those of you who are interested in just how much of a golden ear you truly
are: download Harmon's "How to Listen" software for Windows or Mac OS X
<http://harmanhowtolisten.blogspot.com/> (scroll down).

Harmon requires its trained listeners to pass tests based on this software
before participating in juries to evaluate Harmon products. It doesn't
directly address the sample rate/bit depth issues discussed in the linked
article, but it does address a lot of the issues brought up in the HN
discussion, so you can have a chance to see how much those characteristics
really matter.

You may be surprised.

------
JangoSteve
Even without debating the science and signal processing arguments raised...

 _In any test where a listener can tell two choices apart via any means apart
from listening, the results will usually be what the listener expected in
advance; this is called confirmation bias and it's similar to the placebo
effect. It means people 'hear' differences because of subconscious cues and
preferences that have nothing to do with the audio, like preferring a more
expensive (or more attractive) amplifier over a cheaper option._

 _The human brain is designed to notice patterns and differences, even where
none exist. This tendency can't just be turned off when a person is asked to
make objective decisions; it's completely subconscious. Nor can a bias be
defeated by mere skepticism. Controlled experimentation shows that awareness
of confirmation bias actually increases rather than decreases the effect!_

Doesn't that completely negate his conclusion, that there is no point to
distributing 24/192 music? If people want to pay for 24/192, and even he just
admitted that they will legitimately enjoy it more, how can you conclude there
is no point?

Life is short. I want to enjoy things. Whether or not my enjoyment can be
quantified or scientifically defended, I really don't give a shit. But that's
okay, if you don't want to sell me 24/192 music, Amazon will. Between this and
DRM-free content, it's no wonder I buy all my music from Amazon these days.

~~~
jpdoctor
> _If people want to pay for 24/192, and even he just admitted that they will
> legitimately enjoy it more, how can you conclude there is no point?_

Sorry, no time to reply. I gotta run and write up my biz plan to distribute
32/384 audio.

~~~
joshAg
SUCKER! I'm already working on 48/768 audio. It's amazing how clear the
recordings are.

------
untangle
This article is one of the most lucid and accurate that I have read on this
topic.

However, one thing that's missing here (and in nearly all other similar
pieces) is a full discussion of the prerequisites of the sampling theorem. For
example, the signal must be bandwidth-limited (and no finite-time signal can
be).

But this is a minor concern, as there are many elements in the analog domain
of the recording and playback chains that serve as low-pass filters - starting
with the mics. So bandwidth-limiting is effectively achieved.

For a similar reason, the discussion of the "harmful" effect of high-
frequencies to playback electronics and loudspeakers to be a bit overdone IMO.
Peruse the excellent lab results of modern audio gear on Stereophile's web
site. You'll find that bandwidths exceeding 30kHz are rare.

One last thing. When doing subjective "testing," keep in mind that what some
folks are hearing may be limitations of their gear. For example, most DACs
derive their clocks for higher sampling rates (88/96/176/192) by clock-
multiplier circuits. IOW, 44kHz and 48kHz are the only ones clocked directly
by a crystal. These multiplier circuits are often noisy, contributing to
jitter. The audible effect of this jitter is hard to predict.

Bob

PS As an avid audiophile, I find the clash of subjectivists and objectivists
on this normally-buttoned-down forum to be a bit of a trip.

------
blackhole
You always record stuff at 24-bit/192 kHz for many reasons usually involving
minimizing analog artifacts and to give you a lot of information to work with.
You use 32-bit float wavs to transport stuff around so you don't have to worry
about normalizing levels and clipping. Lossless formats drastically improve
the quality of transients by an enormous degree. But every single objection to
this is either ignoring the points of the article, or talking about the
benefits of recording at high fidelity, when this entire article is pointing
out that once you have _finished a mix_, there is no reason to distribute
things in 24-bit/192kHz. Most speakers can't even play about 20kHz anyway,
which makes the entire point moot. I don't care if you have a bajillion kHz,
the speakers can't play about 20 kHz, so your screwed.

~~~
mfarris
You're getting two entirely different things mixed up.

192 kHz is the sample rate. 192,000 slices per second. It does not refer to
the audible sound spectrum.

20 kHz in speakers refers to the cycles per second of the audible waveform.
Normal human hearing rage is 20 hz - 20 kHz. For most people, it's less than
that.

A speaker can certainly play back music sampled 192,000 times per second. Most
of them can't play tones that are higher pitched than 20 kHz, which is fine
because mostly only dogs can hear up there anyway.

~~~
blackhole
I am not getting these things mixed up, because the sample rate is related to
the maximum frequency that can be stored, and lo and behold, look at all these
people claiming that those higher frequencies matter. 44.1 kHz sample rate can
only encode tones up to about 22 kHz, whereas 192 can encode frequencies of up
to 81 kHz, and those people up there are arguing that these higher frequencies
are exactly why 192 kHz is superior. Now, if you want to say that sampling a
tone at 44100 times per second somehow won't sound as good than 192000 times
per second, I'm not saying that isn't possible, but I don't really take that
claim seriously at all.

The fact is, simply distributing music in lossless format carries the vast
majority of audible improvements. Arguing over whether or not its 24-bit or
16-bit or making a chunk of sound last 5.2 microseconds instead of 22.67 seems
incredibly stupid to me, because you're better off simply improving the mix
itself then fiddling over such microscopic differences. These things only
become relevant if your mix and performance and recording equipment (or
synths) are absurdly close to perfection. This becomes even LESS relevant in
an age of indie-musicians.

------
jwatte
The sampling theorem is for static signals and perfect filters. Turns out,
music isn't static. Once you have transients in the program, you need higher
bandwidth or you will end up with phasing effects (time domain aliasing.) This
is plain from the math!

Filters are also not perfect (but good oversampling filters are not the
weakest link)

Further, even perfectly dithered 16 bit data can't go 20 dB below the
quantization floor, unless you give up on frequency response on the high end.
Again, this is plain math.

With a calibrated 105 dB low-distortion sound system, in a quiet room, I can
hear imperfections from 16 bit, 44 kHz material, especially in soft flutes and
triangle type percussion. Of course, D class amplifiers, and MP3 encoding, do
worse things to the signal, so let's start there. But 20 bit, 96 kHz (or at
least 64 kHz) are scientifically defensible, when analyzing the math and the
physics involved. No snake oil needed!

------
wickedchicken
For an article containing a lot of "well, if you knew signal processing..."
there are two fairly major oversights:

1) Any well-designed system is going to have headroom. Period. Just because
48kHz can capture the frequencies the human hear theoretically, it's always
good to have a little wiggle room. This comes into play even more with
interactive situations: humans are particularly sensitive to jitter. Having an
"overkill" sample rate lets you seamlessly sync things easier without anyone
noticing.

2) 192kHz comes with an additional benefit besides higher frequencies: it also
means more granular timing for the start and stop of transients. More accurate
reverb would be the obvious example. I don't know if the human ear can discern
the difference between 0.03ms and 0.005ms but it's something I don't see
mentioned often.

~~~
Joeboy
> I don't know if the human ear can discern the difference between 0.03ms and
> 0.005ms but it's something I don't see mentioned often

That's the time it takes sound to travel 8mm. Do you think you could tell if
an instrument was positioned differently by 8mm?

~~~
barrkel
The ears distinguish directional audio in part from timing differences in what
hits each ear.

<http://en.wikipedia.org/wiki/Sound_localization> cites
[http://web.archive.org/web/20100410235208/http://www.cs.ucc....](http://web.archive.org/web/20100410235208/http://www.cs.ucc.ie/~ianp/CS2511/HAP.html)
that suggests the brain is sensitive to timing differences between ears as low
as 10 microseconds, or 0.01ms.

~~~
xiphmont
It's not the timing differences, it's the phase differences. The ear is
exceptionally sensitive to phase differences between the ears below 1kHz. This
information is captured exactly (to well beyond the naive precision of the
sampling clock) for any frequency below Nyquist.

------
nileshtrivedi
What I would love to have is: independent instrument/vocals tracks along with
a default recommended "mix". The default mix would be used for normal playback
and independent tracks would be great for custom mix / karaoke etc.

Is this too unrealistic to expect? Has something like this been tried before?

~~~
noisebleed
Trent Reznor / Nine Inch Nails has done it several times:
<http://www.ninremixes.com/multitracks.php>

Plenty of other artists have as well, but this is the most high profile
example I can think of. I agree it would be great if it happened more often.

~~~
polshaw
I'm not a huge NIN fan, but Trent is truly awesome when it comes to digital
music. You can add excellent mastering and dedicated surround mixes too..(rec:
Social network soundtrack). Also a former oink'er.

The beatles multi-tracks are also available (although they were only recorded
4-track so not every instrument always has it's own track), and there has been
a handful of artists who have released their samples of one song for remix
competitions (Daft Punk, Royksopp, Booka Shade).

------
polshaw
I have to say that was probably the most comprehensive dealings with the issue
of sample-rates I've ever come across. I'm not going to make the mistake
others have of claiming falsehoods (all of which i've read so far have been
debunked to my satisfaction by the HN users-- i'm impressed, guys).

As pointed out, mastering has _vastly_ greater effect on the audio quality
(and is often pretty poor[1]), and is the reason vinyl records often can sound
better than their digital counterpart, despite being an inferior
technology[2]. The DAC used also has a massive effect on the sound once you
get into decent quality equipment.

Like the author, i'd also love to see some expansion of mixed-for-surround
music.

[1] a lot because of loudness wars, as pointed out in the post, but also just
due to a lack of time/care/love(/demand?).

[2]
[http://www.hydrogenaudio.org/forums/index.php?showtopic=6175...](http://www.hydrogenaudio.org/forums/index.php?showtopic=61758)
This thread explores the bit-depth of vinyl records, beginning with a claim of
a maximum 11-bit resolution-- limited by the width of a PVC molecule the
record is made from.

------
WalterBright
My hearing has declined over the years, to the point where audiophile gear is
a complete waste of money. For example, I can no longer hear the difference
between a cassette tape and an LP. I still listen to and enjoy music all day,
but no longer worry at all about the sonic quality of it.

My advice to you younger guys is to keep the windows rolled up while driving.
I have no other explanation why my left ear is much worse than my right.

------
Andys
This is a really convincing article that makes me want to set up a double
blind test for myself with my own equipment.

In my own tests I believed that I couldn't tell the difference between 16/44
and 24/96 on high quality loudspeakers, but I could with high quality
headphones. The studies cited all seem to use loud speakers in testing.

Also worth noting, the article states that obtaining 24/96 source material
sometimes means you get better mastered material, which still sounds better
after down-sampling back to 16/44.

~~~
hackermom
You weren't just believing things. The difference between 44khz and 96khz
sample rate is very noticable even with mediocre audio equipment. It's an
overstatement to refer to the situation as a "hi-fi case". 16/24 bits however
makes no difference at all except on the size of the material.

------
leouznw
I know a bit of sound engineering, waves and so.. I totally agree with the
title and the first 60 lines of article, and I add my POV: 1\. Most of the
people doesn't care, 2\. What apple did is just about marketing, 3\. Most of
the people who says that care is pretending, 4\. Zeppelin still rock the shit
in a poor quality mono mp3 recorded by a drunk guy in the audience of a
concert in 73.

~~~
kokey
I do care, but I'm not the average user. Apple has always catered well for
those in audio and video, up to professional levels. These are markets that
retain Apple users, even when Steve Jobs was between Apples. It seems like
Apple is only requesting masters to come in a higher resolution, not that
consumers will generally end up with these. I think this is entirely fair
since before you want to modify something (e.g. to remaster it for iTunes) you
want to start off at a good quality high resolution.

That said, if Apple also allows high quality recordings to be sold, it will be
useful. For example of their acapellas, instrumental tracks or samples, it
would be convenient for others who want to want to remix it, and iTunes would
be a platform for this trade.

Also for tracks DJs play. Most compression throw away a lot of the bass which
people can't hear, but this is bass you can feel rumbling through your guts on
a big sound system and is part of the experience.

For the rest, they were happy with low rate AAC files on the early iPods, they
are happy with the sound coming from their crappy little iPod dock, for them
it won't make a difference as long as it's a chart music track from a
memorable and impressionable time of their life.

------
jlft
In normal listening conditions and for most people the difference between
16/44 and 24/192 is inaudible.

Given a 5 minute song, if I have the choice to download a 11MB file (320kpbs
MP3) or a 330MB file (24/192) I would of course choose the 11MB file. The
sound quality is perfectly acceptable and the file size much more convenient
to manage (storage, backups, etc.).

In terms of the convenience of managing the file size and sound quality I
think 320kbps MP3 is the best compromise.

Here's a file size comparision of a 5 minute stereo song:

MP3 128kbps > 5 MB

MP3 320kbps > 11 MB

Uncompressed 16/44 > 50 MB

Uncompressed 24/192 > 330 MB

When talking about sound quality there is a much more relevant issue: the
amplitude compression (distortion) abuse used by mastering engineers and
producers that totally destroys the dynamic and life of the sound. That is a
real issue. When buying a song there should be two versions to choose from:

A) "Loud", dynamically destroyed / distorted version.

B) Normal, dynamic, non-distorted version.

Today only version A is available to buy.

~~~
rdtsc
But then for every 10 people like you there is 1 person who is willing to pay
20x as much so they can get a "higher fidelity" product.

For a producer and manufacturer the rational approach would be to cater to
that craziness and extract as much money from it as possible. In other words
if you are selling HDMI cables, spend $2/cable to make it, then sell most for
$5 and then re-brand some and sell for $500. If only takes 1 out of 100 people
to buying that to make the same profit. You know these people are obsessed and
irrational so you cater to that. And that's basically how we end up with
ridiculously overpriced Monster cables and recordings distributed to customers
@ 192kHz.

~~~
jlft
Agreed, that market exists. My point is, why discuss the subtle difference
between 16/44 vs 24/192 when there are far more audible and damaging practices
going on in the music industry. For example, aggressive compression and brick
limiting which adds distortion to achieve maximum loudness ('loudness wars').

------
blahblahblah
I mostly agree with the article in the context of distribution of a final mix.
However, the article ignores one glaringly obvious reason to distribute in
24/192 format: to allow the listener to be a participant in the creative
process, enabling better results for amateur musician listeners who want to
sample or remix the audio or for DJs to get better results when altering the
tempo for beat matching one track with another, etc. Of course, if you're
going to do that, you might as well distribute in a multi-track format instead
to maximize flexibility for the end user (Want to sing karaoke? Just turn off
the lead vocal track for playback).

~~~
nullc
Yea, and and if the bandwidth/storage is at all an issue 6x size bloat from
24/192 pays for 6 separated tracks. (Actually more, because multitrack is more
losslessly compressible while 24/192 is less). If you're already providing
multitrack then 24 bit audio would make sense... otherwise, meh.

------
jdc0589
There is no harm in releasing higher quality uncompressed or loss-less tracks.
At the worst they will bring in some new customers, such as myself, that
currently will not buy music online. Why would I pay $10 for an album as a
highly compressed download when I can pay the same price for the CD and rip it
to FLAC myself? I realize I am in the minority here, but as CDs phase out even
more, there has to be some other way for consumers to obtain high quality
versions of tracks.

Footnote, you don't have to have a >$10,000 setup to benefit from higher
quality tracks (compared to the downloads that sometimes have 'questionable'
quality). I have two systems, a full range stereo (front left and right) setup
for nearfield listening at my desk thats +/- 1DB from 50hz-20khz. The other is
a stereo setup in my media room; 2 way quarter wave transmission line, +/-3DB
40hz-20khz. The point is, there are a lot of people with less than $1200 in
audio gear that still want lossless tracks made available. Who cares if the
human ear can't discern _much_ of the extra information, we still want it.

------
neilalbrock
A few years ago I became really interested in recording music. I had been
writing a little with a friend, using whatever crap equipment we could afford,
the results weren't amazing but we were having fun and staying focussed on the
music itself.

Then we starting recording other people. I became obsessed with gear, software
and all the associated toys that go with any technical pursuit. I'm a
programmer, so it's easy to understand how that happens but I totally lost
sight of the music, spent way too much money and equipment that was nowhere
near being required and generally lost the plot. I was tracking everything
24-bit/96kHz and bemoaning the loss of quality when I mixed down for CD.

Anyway, the TL;DR version of what followed was that we recorded quite a bit,
lost interest in making our own music and then the whole adventure came to an
end. Now my gear is leaving via eBay and I'm finding my way back to just
playing guitar and trying to write good music.

24-bit/192kHz - pointless. Give me a small venue and a guy with an acoustic
guitar any day.

------
zzygan
This is a good article, however the guy who has been pushing this for years
and years now, is a man called Dan Lavry. In fact he wrote a very good,
rigorous explanation a few years back,in very readable and well written form.

[http://www.lavryengineering.com/documents/Sampling_Theory.pd...](http://www.lavryengineering.com/documents/Sampling_Theory.pdf)

------
rbanffy
Minor nitpick

> The FLAC file is also smaller than the WAV, and so a random corruption would
> be less likely because there's less data that could be affected.

At the same time, if you flip a bit on a WAV file, you may hear a "pop" sound.
On a FLAC file, the whole encoding block may be inaudible (or worse).

------
yzhou
The hearing of ears is a time-domain thing, not a frequency domain thing. It's
the frequency response of all the frequency components added together. people
might not be able to respond well to a single high frequency tone, but might
respond well to a combination of tones.

~~~
xiphmont
No. It's both.

The basilar membrane is a loosely tuned resonator. The hair cells placed on it
fire beginning on the positive zero crossing. So, to a first approximation,
the ear is in fact a filterbank.

There is a time domain component in that the cochlear nucleus contains nerve
cells that watch multiple hair cells at a time and correlate the firing in
several different ways. Some attempt to discriminate pitch, some convolve and
correlate in-phase firing energy, some look for tones to end, etc. This
information is then forwarded on to the brain.

However, getting back to your point, no hair cells will fire if the basilar
membrane doesn't move, and it's tuned to a frequency range.

------
tammer
I find mp3 and aac compression artifacts to be monstrously irritating. I have
no idea how the majority of the world seemingly can ignore them.

Further, I can hear a difference between 44.1kHz and 96kHz. Whether you can
hear that difference is up to you. (The word-length is a red herring - there's
no new information contained in a 24-bit recording vs 16.)

IMO anything less than flac and you're missing something. Higher sampling
frequencies do add to the sound, but in a way that is almost invisible to the
untrained ear. Perhaps these should be distributed at a premium the way SACDs
and similar "audiophile" formats were in the past?

------
sliverstorm
So, presuming we take this example:

<http://people.xiph.org/~xiphmont/demo/jaggy2.png>

The key to reproducing the original signal from the digital signal is a low-
pass filter that rejects everything above the sampling rate, correct?

That is to say, what I am getting at is while the original signal can be
reproduced, it requires properly tuned, and probably reasonably high
performance, hardware to remove the higher frequency components of that square
wave. Can you count on consumer grade hardware to do this well?

~~~
nullc
Yes, thats basically it. They do this _exceptionally_ well in fact.

Typically the technique used inside DAC is to digitally upsample the signal
(by duplicating samples, often to a few MHz— also allowing them to use a low
bit-depth DAC) then it applies a very sharp "perfect" digital filter to cut it
right to the proper passband (half the sampling rate). The analog output then
contains only a tiny amount of ultrasonic aliasing which is so far out that
it's easily rolled off by simple induction in the output.

This isn't just theory. Here is a wav file I made at a 1kHz sampling rate,
where every other sample is -.25/.25: <http://people.xiph.org/~greg/1khz-
sampled.wav> (so a 500Hz tone, the highest you can represent with 1kHz
sampling).

Feeding that file to a boring resampler (I used SSRC, but anything should give
roughly the same result— a least when not quite so ridiculously close to
nyquist, most will attenuate near-nyquist data extensively) and get this:
<http://people.xiph.org/~greg/1khz-sampled-to-48khz.wav>

Here are the two signals plotted against each other:
<http://people.xiph.org/~greg/1khz-to-48khz.png>

As you can see— the 500Hz sinewave is reconstructed perfectly. (Of course, a
500Hz square wave would not be (you'd get a sinewave out) but this is because
a 500Hz square wave contains energy far beyond the nyquist of 1kHz sampling).

Here is a spectrograph of the same signal <http://people.xiph.org/~greg/1khz-
to-48khz-spec.png> showing that the tone is indeed pure (the faint background
noise is the dither the resampler applies when requantizing its high precision
intermediate format back to 16 bits).

------
noonespecial
I was under the impression that two inaudible high frequency tones could
interfere with each other to create an audible interference pattern. (I think
known as a "beat frequency").

If this is the case, then all of the arguments in the world about the maximum
audible single frequency are irrelevant. Imagine music composed entirely of
these beat frequencies and performed with a pair of oscillators between 25kHz
and 35kHz. Without higher resolution encoding, it would be audible IRL but the
recording would be silence.

~~~
zb
If the beat frequency is audible, it will be on the recording. Obviously.

~~~
noonespecial
That would suppose that the recording device precisely matched the orientation
of the listener, and the recording was not created digitally in (multi-track
fashion for example). There would have to be air space in order for the
interference pattern to set up in.

So you'd be right if your mics were head spaced and in the venue. But you'd
still have secondary data, with the original lost.

~~~
ef4
> But you'd still have secondary data, with the original lost.

By that standard, the original is _always_ lost unless you have a completely
holographic recording. 192kHz doesn't help with that problem at all.

------
ChuckMcM
TL;DR - long and detailed information about why if you got music in 24/192
format you couldn't tell the difference between it and 16/48 music.

I chuckled because this is so true, and yet tell that to the people who buy
oxygen free copper 'monster' cables for their speakers, being careful to align
the arrows with the direction of the music from the amplifier to the speaker.
People, even otherwise reasonable people, will swear up and down they can hear
the difference.

------
yzhou
A person can not hear a 22kHz tone doesn't mean he can not hear a sound that
contains 22kHz components. For example, a square wave contains lots of high
frequency harmonics, the more higher frequency harmonics it have, the
"squarer" the square wave gets. An ideal square wave forms ideal "0" "1"
states. A person's ear might not be able to hear a 22Khz sine wave tone, but
he might be able to sense the steepness of "0" "1" state.

~~~
nullc
First, if you can "hear the steepness" you really can hear higher frequencies,
but I assume you meant "maybe you can hear higher frequencies but not higher
frequency _tones_"...

People have suggested this. It's been tested in rigorous double blind tests—
involving both real music signals as well as special test tones (Linked from
the article). The tests were unable to show that people could hear the
ultrasonics. Moreover, there isn't any physiological basis to expect people to
be able to. You can't expect a stronger result than that.

Common 48KHz audio already goes a bit beyond what adults are known to be able
to hear, so you've already got some headroom for "but what if a few people
hear better than anyone the researchers have been able to find!".

------
agentgt
I know this is slightly tangential but are hi-end DACs really worth it? I have
always been amazed how much audiophile DACs cost ($300-1000). The reality is I
listen to 320kbps music that was most likely recorded at 44100. DAC technology
is not exactly new. So why the price?

Another tangent: To me it seems audio engineering should fix the "woofer".
That is it seems subwoofers have terrible distortion.

~~~
TylerE
A low-end dedicated DAC is likely to be a substantial upgrade over a built-in
soundcard (I'm assuming we're talking PC sound here). A PC case is a pretty
noisy place, electrically - I know one one work PC I had once you could
actually here the mouse move, if you had heaphones on and cranked the volume
with nothing playing - horizontal and vertical movement had different
frequencies.

The move from a low end ($150-300) DAC to one much more expensive will be
considerably less drastic, and likely won't matter until you've dropped at
least $5k in to the rest of your system.

That said, you may already own a DAC without realising it...as long as you're
taking the singal out _digitally_ (e.g. SP-DIF or digital coax) to an external
receiver, you're already in a pretty decent place.

~~~
agentgt
Oh yes I agree an off board DAC is better. I own the Fiio E7 which I highly
recommend for laptops and only costs $80.00. In fact I run a 50 foot USB cable
from my laptop to my DAC and the improvement is much better than running a 50
foot 3.5mm TRS.

But the high end ones that are 24 bit 192khz that cost $1k (Cambridge
Soundworks DAC magic comes to mind) I have to seriously doubt I'm going to
hear it. I really only hear the DAC difference (compared to my laptop and
FIIO) when I use headphones.

------
tcarnell
Has anyone had a look at their hi-fi amp recently? If probably probably
doesn't handle much more than 80 kHz and your speakers probably dont respond
to anything over 20 kHz. So yes, 192 kHz is pointless UNLESS you intend using
it for studio quality editing/mixing - and I'm sure Steve Jobs would not have
encouraged this!

------
yu
From Footnote 1: [...] If we were to use the full dynamic range of 24bit and a
listener had the equipment to reproduce it all, there is a fair chance,
depending on age and general health, that the listener would die instantly.
The most fit would probably just go into coma for a few weeks and wake up
totally deaf.

------
jaekwon
The article AFAIK states little about distortions introduced in remixes &
samples. I would expect certain high frequency samples, when mixed together to
overlap in time, would introduce moire artifacts (beats).

~~~
nullc
Not unless you pass them through a non-linear filter like a distortion effect.

(yes, the ear-brain system is non-linear too— but apparently it filters out
the ultrasonics before they do anything measurable in this regard)

------
bryanlarsen
One of the strongest things that makes this article credible is that in it we
have the author of Ogg Vorbis recommending that we stop using Ogg Vorbis (and
all other lossy compression formats).

------
thewisedude
I am told that a similar argument can be made between TV's that display at 120
Hz as opposed to 240 Hz. i.e there is no discernible difference!

------
jensnockert
I just want floating point, then this silly loudness war would end (to some
extent, since you can make the mix almost infinitely loud).

~~~
FrankBooth
It would make no difference at all.

------
tintin
I think this only applies to headphones. People also 'hear' sound with there
body (skin). Maybe you could call it experiencing sound. And then there are
resonating sounds that cannot be heard but help to create other sounds. But
maybe this won't apply to a recording because your will record the result and
not the tones that make the result.

This is a great article but I'm still not convinced people cannot have a
sensation of sound out of there hearing range.

~~~
tintin
Never mind. I read 192kbps instead of 192kHz. 24 bit might have some
advantages but 192kHz not.

------
mistercow
> Can you see the LED flash when you press a button? No? Not even the tiniest
> amount?

I used to be able to see it when I was a kid (it looked very faintly red), but
I just tried it and couldn't see it at all. That's actually a little bit
disturbing.

~~~
xiphmont
I would guess that's because some earlier remotes used a higher frequency IR
emitter that was in fact touching into the red.

These days, the various IR communication protocols have been standardized and
virtually all use 920nm, 940nm or 980nm emitters, all of which will be
invisible. I mentioned the Apple IR remote specifically because it's a remote
most people reading TFA will have, and it's known to be a 980nm emitter.

------
diminish
Would someone explain should I use 44.1 or 48Khz?

------
naughtysriram
I think 192kHz is the sampling rate used by the A2D converter vice verca. It
is not the actual frequency of the sound (data).

------
hackermom
There is no point with going over 16 bits, but there is definitely a point
with going over 44.1khz, as it allows you to actually reproduce waveforms more
accurately than 44.1khz. Try reproducing f.e. a sinewave accurately over
4-5khz with a sample rate of just 44.1khz - it cannot be done, and at this
point we haven't even taken into account the issue of varying slew-rate
characteristics of the thousands or so different DAC output stages in use in
personal audio equipment.

44.1khz gives too much aliasing distortion, but 192khz is quite the overkill.
Ideally, digital audio could sit on 16 bits of depth sampled at 96khz.

~~~
nullc
No. This really is not the case. The article _specifically_ addresses this
misconception.

The signal reproduced from your 44.1kHz sampled digital input is not a stair-
step like some broken waveform editor might display: On output it goes through
a matched reconstruction filter (which may, in fact, be digital and involve an
oversampled DAC or it could be analog though those are harder to build without
compromise). After the reconstruction filter the output is _EXACT_, assuming
the input only contained energy below the nyquist (well, and was sufficiently
far away from the reconstruction lowpass).

So even a 5khz sine wave is reproduced perfectly with 44.1kHz sampling.

~~~
mturmon
@nullc: of course you're right, and the commenter you're replying to does not
understand the Nyquist-Shannon sampling theorem. Which is a shame, because the
article specifically addressed this point.

These discussions of audio standards always get sidetracked by people who
don't understand or believe this result. (Have to admit, the result _is_
surprising).

I think there may be problems with the argument in TFA, which is based
exclusively on standard linear systems theory.

Of course, the ear and some of its perceptual components may be significantly
nonlinear, and thus not covered by the frequency response graphs of TFA.

These graphs assume linear systems, in which you put two frequencies in, and
the same frequencies pop out in scaled form. Nonlinear systems can produce new
frequencies in response, and this possibility is not discussed in TFA.
Probably these effects are quite minor, but may be audible to some listeners
on some equipment for some choices of source material.

~~~
nullc
Indeed, but if there were non-linearies in the ear (there are many, of course)
which allowed detection of ultrasonics (less likely, because the first stage
of the ear is impressively linear) you'd expect them to show up in the actual
listening tests.

The TFA does at least make this the-proof-is-in-the-pudding point somewhere in
its depths. :)

------
joccam
Sometimes less is more. The debate goes on. Why not just let the music play?
And by that I mean high resolution music. All you need is one person who can
hear high frequencies, and all the technical mumble-jumble becomes hogwash.

People actually _believe_ the 20KHz argument that anything above is inaudible.
That's hogwash. I know because I can hear (or sense) higher frequencies, and I
do not have the absolute best ears I've ever "met."

For example, last week I attended a A/V equipment event with very high-end
equipment. It was packed --- over 600 people for one evening. 6 rooms of
equipment. I'm sure all six served the same fare according to the 20-20KHz
argument of this piece, yet they all sounded quite (or even extremely)
different.

The 20 KHz argument is a myth. For people who can't hear the difference, no
problem. But please do refrain from ruining or hobbling music for the rest of
us... who can hear a wider frequency range.

Yes, some people are color blind. Does that mean the rest of us shouldn't use
color? I hope not.

Music is an important wholesome and potentially emotional part of human life.
Please do not cap it with "false optimizations".

24-bit/192 KHz is not inferior to CD quality sound. If you don't believe me,
try a Linn system sourced on a Klimax DS with some high bitrate Linn classical
music (or the Beatles Masters USB release!). If you can't hear the difference
compared to low bit-rate (including CD quality) material, I assure you someone
can. The low bit-rate will sound flat, hollow, less lively, or/and more
coarse. Any number of problems exhibit at inadequate bit levels.

Vinyl is analogue quality (no discrete digital distortion). CD quality is a
large step down from vinyl. A/V is just trying to get vinyl like quality from
digital. We don't need nay-sayers impeding progress. If you can't hear the
difference, please let someone who can hear make the informed decisions.

Thanks.

~~~
_delirium
It's not a myth, but a fact established in laboratory studies. Your anecdotal
claims to hear frequencies that scientific evidence suggests you cannot hear
doesn't overturn science. I'd be convinced if you correctly identified which
speakers were reproducing 21 kHz frequencies in a double-blind test, though.

~~~
joccam
Isn't science verified through (wait for it...) experimentation? So how does
my hearing not invalidate your science?

That's the problem with the theoretical science. When it's false, it's false.
Come up with a new hypothesis; this one's false as it pertains to human
hearing. There's information theory, and then there's auditory reality.
Reality confounds the theory as applied to hearing. I don't know where the
fault lies, and I don't really care.

But it's really annoying and frustrating having people nix progress out of
idealistic theory, "laboratory" studies, and ignorance. The experiments (my
experiences and numerous others) don't lie.

Double-blind is great, but I can already tell the differences between all six
rooms of equipment from last week. One of the rooms was so extreme, I wanted
to run out of the room due to discomfort (but I was polite and stayed all 30
minutes). In other words, double-blind was unnecessary. Someone whose ears I
respect a great deal, loved that room. Even golden ears don't all hear the
same. But I don't need double-blind to confirm trivial experience. The proof
is already in the listening.

~~~
_delirium
> So how does my hearing not invalidate your science?

Because it's not a blind study. In audio, claiming something sounds better
than something else is low-strength evidence, because it doesn't: 1)
distinguish psychological bias (which is very strong in this area) from actual
audible results; or 2) distinguish which characteristics of speakers, if any,
you may be hearing.

If you can consistently ABX two speakers that have similar characteristics
except that one reproduces frequencies over 20 kHz while the other doesn't
(with identical performance below 20 kHz), I'd be convinced. One possibility
is to use the same speaker but insert a high-quality 20 kHz lowpass in the
chain during part of the test; or use the same speaker but with 44 kHz versus
96 kHz source material. I've never seen a controlled, blind case where a human
can tell the difference there.

~~~
joccam
The psychological component is a red herring. Even though I already have a
system (bias), I don't care about the other systems. I went to the show for
enjoyment, education, and appreciation. Some of the systems were unknown to me
(no bias), and some were known and surprised me in some ways (again, some bias
overridden). So bias can be important, but it's not relevant in this case. So
bias doesn't invalidate my experience.

As for the double-blind and high frequencies, I believe I've already done the
test. I have had my hearing tested several times. One of them, I recall the
tester actually asked me to repeat some tests... it was funny. The testing was
at very high frequencies. I believe she thought I was guessing the
higher/lower frequencies... and getting lucky. So (I strongly suspect) she
wanted to "prove" to herself what you want to prove --- that noone can hear
above 20KHz. I disappointed her. I think she even threw in some placebo tests
(no frequencies at all). It was funny. She never explained herself. I suspect
she just thought I got lucky again.

How to really test this stuff? Get one of the audio designers to test... but
they will laugh in the testers' faces. They do this stuff for a living... to
build real products... for real live customers who can hear the differences.
Dave Wilson was at the A/V show. Try listening to a pair of Wilson Audio
speakers. I bet he can hear better than just about anyone... His speakers
(when sourced and driven properly) are that good. But he wouldn't waste his
time on such tests. He has customers to serve and a business to run.

I doubt lab experiments look to disprove their theories once and for all.
That's a social prejudice built into the lab experiments. Fix that, and you'll
end up with a better hypothesis.

------
rbreve
Unless you are a dj or producer and would like to sample or time stretch the
tracks. That's why Beaport offers a wav download option, that many
djs/producers prefer.

------
aiurtourist
Science be damned! Onwards with subjectivity!

• 24-bit audio is magical. When I recorded myself playing guitar in 24-bit and
played it back through my amp, it sounded like I was still playing. 16-bit
sounded like a CD.

• With MP3s, 192 kbps is a huge step up from 128 kbps. 192 doesn't exhibit any
of the "swooshiness" heard in the upper range of 128 kbps MP3s for regular
rock/pop/hiphop music.

~~~
danbmil99
I believe you are confusing 192kbit compression with 192 khz sampling rate.
Not at all the same thing.

~~~
aiurtourist
Whoops! Guess I'll read more carefully, then.

------
hackinthebochs
One thing I don't see addressed is the experience of _feeling_ frequencies
that can't directly be heard. There was a study done with a particular piece
of classical music, with and without a particular inaudible component to it.
The presence of the inaudible component drastically changed the listeners
perception of the music. They described it as more dark or creepy (perhaps not
the actual words used, but it matches the sentiment). The point is that there
may be value in reproducing frequencies that we can't "hear", as inaudible
notes can alter the experience of the music.

*not the study I was referring to but its along the same lines: [http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=5291...](http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=5291285)

~~~
notahaxor
The author completely ignores infrasonics and writes under the incorrect
assumption that our only perception of wave pressure comes from our eardrums.

I've never been able to enjoy listening to my favorite classical music on
headphones or even smaller speakers, and it's largely because of the effect
you describe.

At this point I'm resigned to preserving my treasured (and cumbersome) vinyl
collections. Maybe if Apple comes up with some snazzy marketing term (e.g.
"Retina") for 24/192 or even 24/92, and starts distributing it on iTunes,
things might start to change.

~~~
mdarens
You don't need a higher sample rate to capture or play back infrasonic
pressure waves, but most recordings are mastered to remove DC offset and
rumble <20Hz, as reproducing those components requires specialized equipment,
such as a rotary subwoofer.

~~~
notahaxor
Wouldn't the higher bit rate help though? I've long suspected my preference is
largely due to the difference in mastering techniques more than any technical
limitations.

------
citizenspaced
I don't understand why anyone gets down on 24-bit consumer audio.

Specifically because CD-quality 16/44 audio has midrange distortion present
during complex passages that is completely eliminated and non-present in 24/96
sources.

Listen to "Us and Them" off a 16/44 CD version of the Pink Floyd album Dark
Side of the Moon. When it kicks into the chorus, it becomes totally distorted
and everything in the midrange bleeds into each other. It's a mess.

Then, try listening to the 24/96 Immersion box set copy or a vinyl-sourced
24/96 rip and you'll find it's gone. When the song gets complex and loud,
everything remains totally clear, each instrument stands on it's own, it
doesn't become an awful distorted jumble.

You could argue that it's just the quality of the master that makes the
difference; but if you take a copy of the original transcoded to 16/44 and
compare it again with the 24/96 copy you can hear the same effect.

Why would anyone argue against high-resolution audio anyway? Sure, most
everyone will probably just continue downloading 16/44 MP3s, but at least give
us the option to have 24bit FLACs of the stuff we really like. Please and
thank you.

~~~
dedward
You could argue it's the quality of the master, and the mastering process, and
you'd be right. That's a no-brainer.

"but if you take a copy of the original transcoded to 16/44 and compare it
again with the 24/96 copy you can hear the same effect." I could believe that,
but do you mean to do the transcoding yourself? IN this case you become the
engineer, and the tools you use and all that become vital as well.

Having heard stunningly awesome CD's of DSOTM on a homebuild heathkit amp and
some old speakers and not believing my ears when I saw what the setup was, I'm
skeptical... can't help it.

------
coopersloan
Huh, I think people truly advocating 192 as a distribution format will be few
and far in between, a really good and cheaper sampling system can be put
together at 96. Still, a lot of things in this article perplex me.

Human hearing is limited to 20k because frequencies higher than that are
perceived as painful? Dont agree with that one.

24 bit doesn't offer any advantages to sound quality? Sheesh.

And the crux of the argument is intermodulation distortion increases when you
try to represent more frequencies? Isn't that an argument for a faster power
amp?

~~~
masklinn
> Human hearing is limited to 20k because frequencies higher than that are
> perceived as painful? Dont agree with that one.

You misread the article. It's because there is so little response that being
able to hear it would blow your eardrums (and even then, it might _still_ be
beyond your ability to hear it). There's no value in that.

> 24 bit doesn't offer any advantages to sound quality? Sheesh.

Not quite what TFA says. According to the article, 16 bits effectively covers
the dynamic range of human hearing, so more than that is pointless _for music
consumed_ by human beings (hence all the stuff about 24bit being a good idea
for mastering & production). If you're storing integers in the 0~16384 range,
going from 16 bit integers to 32 bit ones is not going to give you "better
ints", it's just going to waste 2 bytes per int. Same thing here.

~~~
coopersloan
I can admit that I misread the article when it comes to hearing limits. I was
reacting to my perception as an audio engineer that a lot of people dismiss
the importance of that frequency range.

~~~
lwat
With good reason!

