
Does pono's 24-bit audio matter? Hear for yourself. - hendriks
http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.html
======
rrrooss
This test is flawed in a few ways. The and the error lies in the source
material. The why would there be a perciveable difference between the same
inferior input shown to both encoding formats. Changing between encoding
formats is a topic of its own but the problem here is similar to getting a
640x480 compressed jpeg and putting it into a 1920x1080 lossless format and
the same source into a 1280x720 lossless file and saying. What's the
difference? They both look crappy! Many people have an opinion on high res
audio formats but they are so many places to go wrong. Audio should be given a
little more credit for it immersive ability. When done right it can take you
places. Something to consider is that our eyes can only see 1 octave of
information our ears can hear 10 octaves. Whilst these high res formats aren't
needed for every application they do make a big difference when the source
material can take advantage of it. I feel in 2014 music should be released in
the best format available from the studio and if u want a crappier version so
u can shove 5000 songs on your iWhatever that's your choice.

~~~
microcolonel
It's not the same as the same inferiour source material.

When you apply negative gain to the 16-bit signal, it has less amplitude in
absolute terms, but it should have all the resolution of a whole 16-bit sample
in that space.

The point is that whether that 16-bit signal is represented with the lower 4
bits of a 16-bit sample, or the lower 12 bits of a 24-bit sample, you can
scarcely hear anything at all, let alone a qualitative difference between the
signals.

192KHz is not a higher quality format, it is a production format. The purpose
is to reduce the cost and finality of anti-aliasing filters when sampling the
signal, it is just cheaper to manufacture. There's a good reason for this
being the standard in professional audio, they need to buy a LOT of audio
interfaces, many studios will have thousands of such inputs.

Thanks to the basic laws governing signals, we know with near certainty that
not only is 192KHz overkill, but so is 48KHz, and so is 44.1. Without
significant new evidence showing humans hearing signals with frequencies
greater than 24KHz, you will not make any convincing argument as to why we
should go with any sample rate higher than 48KHz for human listening.

As for 24-bit, it is another production interchange format. It's there so that
you don't need to stand around adjusting gain knobs on audio interfaces so
that you get decent fidelity but also don't clip. With 24-bit you can just
sample your audio once, and assuming it's within a reasonable range, you can
adjust the gain in the discrete signal. There is some indication that 16-bit
sampling is less than completely ideal. The very best ears in humanity(newborn
ears) distinguish about 21 bits in the safe ranges of amplitude, 24-bit may
make sense in an audio system for newborn babies.

I feel that in 2014, music should be released in the best format available
from the studio, and if you want a crappier version so that you can shove only
100 songs on your iWhatever, that's your choice.

------
coldpie
This test is pretty uninteresting, but I'm glad the author linked to the Xiph
article because it's one of the best articles on the subject I've read.

[http://people.xiph.org/~xiphmont/demo/neil-
young.html](http://people.xiph.org/~xiphmont/demo/neil-young.html)

~~~
twobeard
The author admits that "16 bits does not quite cover the entire theoretical
dynamic range of the human ear in ideal conditions". But then the author
concludes "let's not use 24 bits anyway because it doesn't really help anyway,
and it wastes space"

Wastes space? In an age where we stream gigabyte movies? Do we need to remind
the author that we're no longer in 1982 and that we have evolved beyond floppy
disks?

~~~
joshstrange
>> Wastes space? In an age where we stream gigabyte movies? Do we need to
remind the author that we're no longer in 1982 and that we have evolved beyond
floppy disks?

Clever way of leaving out just how much space it takes up (takes up 6 times
the space [0][1]) also as both this author and Xiph concluded you can't really
hear the difference. When we stream HD (As you put it "Gigabyte movies") we
are getting a clearly superior product. I can easily see the difference
between 480/720/1080 whereas I seriously doubt I can hear the difference
between 16 and 24bit audio. Yes space is cheaper than it used to be but it's
neither free nor unlimited, not to mention the primary way people listen to
music nowadays is probably on mobile devices where space is still at a premium
or they enjoy their music via a streaming service a la Spotify/Rdio/Google
Music where bandwidth is at a premium (And storage factors in here as well due
to either being able to store less songs locally or in cache due to larger
file size).

[0] [http://people.xiph.org/~xiphmont/demo/neil-
young.html](http://people.xiph.org/~xiphmont/demo/neil-young.html)

[1] [http://blog.beatunes.com/2014/04/does-24-bit-audio-
matter.ht...](http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.html)

~~~
ygra
The six times figure is for 192kbps _and_ 24 bit. The move from 16 to 24 bit
alone would just result in an increase of 50 %.

~~~
joshstrange
Do you have a source for this? I was under the impression that:

    
    
        filesize(192kbps + 16bit) = 6*filesize(192kbps + 24bits)
    

Also the point of both articles is that: You can't hear the difference.

Look at MP3 vs WAV (or FLAC), MP3 won out (for consumers) because of it's
filesize being so much smaller. Look at the compression differences:

>> Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2
kbit/s,[note 2] so the bitrates 128, 160 and 192 kbit/s represent compression
ratios of approximately 11:1, 9:1 and 7:1 respectively. [0]

At the best quality listed here (192kbps) MP3 is 7 times smaller than it's WAV
counterpart. I just don't see people paying for 6x the space for something
they can't tell the difference between. If anything history has proven this
not to be the case.

[0]
[https://en.wikipedia.org/wiki/MP3#Bit_rate](https://en.wikipedia.org/wiki/MP3#Bit_rate)

~~~
ygra
Trivial arithmetic:

    
    
        192 ÷ 48   = 4
         24 ÷ 16   = 1.5
          4 ×  1.5 = 6
    

The six times is also on the wrong side of your equation, I guess.

But if you need a source, I can cite what you cited:

> 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.

They were talking about both sampling rate _and_ bit depth.

Oh, and just for the record, I do agree with the general idea that it's
useless to ship audio to consumers as 192/24, for the same reason it's
unwieldy to publish photos online as 20+ MiB raw files (even if browsers could
display them). I was just noting that the size comparison included both
sampling rate and bit depth and not just one of them.

------
acomjean
I have pretty good headphones (Sony MDR-V6). I can hear decently enough
(although getting older). Like I noticed that the Deutche Gramaphone 111 years
music collection I bought (256kbps mp3s) has taped performances from the 70 &
80s which have some tape his.

Honestly as an end user its really good enough. If I was remixing this stuff
and processing the audio I would want the non-compressed. (similar to shooting
Raw photographs ). But at 256kbps honestly it really is amazingly good.
thinking back to cassette tapes and records, its astonishing.

As for the tape hiss on the classical music, the performances are so good I
just ignore it. Sometimes I think people get worked up about the quality of
the recording and ignore the quality of the performance.

~~~
calinet6
> Sometimes I think people get worked up about the quality of the recording
> and ignore the quality of the performance.

This is the main problem with audiophiles.

Improve your equipment until it makes your music awesome, then just enjoy it.

~~~
danbee
"Normal people use their equipment to listen to music. Audiophiles use music
to listen to their equipment."

------
tmikaeld
It's not only the difference (or none) between 16 or 24-bit, but it's mostly
about the quality of the DAC that is being used - in this case a very high-end
one.

Then again, it's been proven time and time again that the average person can't
even hear the difference between mp3 or FLAC.

Considering you'd also need proper high-end monitor headphones, i believe this
is more about marketing, the brand and the pitch than it is about what you
actually get or can actually use.

\+ it can only hold about 2000 FLAC files even with expansion card.

~~~
Shivetya
without the big name attached I doubt it would have had much traction. Its
merely a very good marketing exercise. Snake oil for the 21st century. No
better than all the joke techies used to make about HDMI cables.

I always like Alan Parson's quote, audiophiles use your music to listen to
their equipment

~~~
ANTSANTS
A 24-bit sample depth is not completely useless and silly in the way that gold
plated HDMI cables are. You _can_ get a real improvement in dynamic range with
24-bit samples, allowing a song to have quiet parts and loud parts with no
loss in quality in either, which is hard to do with 16-bit samples without
losing quality for the quiet parts or using dynamic range compression and
having everything sound roughly the same.

Of course, few people actually want impressive shifts in volume in their
music, because prog is dead and no one wants their headphones to suddenly blow
their eardrums out. Gotta compress the shit out of that dubstep for the
kiddies! Somewhat ironically, greater bit depth in audio makes a bigger
difference for film than most music because of this.

------
jrabone
One thing that is alluded to but perhaps not explored fully is the use of
noise-shaping dither techniques which rely on pushing the noise out into the
ultrasonic region. Xiph mentions it in the context of 16 bit resolution, but I
believe it interacts with the discussion of 192 kHz sampling.

Significant noise power at ultrasonic frequencies (or even a skewing towards
higher audible frequencies) represents a problem for tweeters & amplifiers
both in terms of the intermodulation distortion mentioned in Xiph's article
and in terms of power dissipation at the business end. IIRC Dell and VLC are
having a falling-out because VLC's soft clipping is damaging the speakers in
certain Dell laptops.

~~~
sp332
Here's the discussion on that Dell issue.
[https://news.ycombinator.com/item?id=7205759](https://news.ycombinator.com/item?id=7205759)
"Simply said, the sound card outputs at max 10W, and the speakers only can
take 6W in, and neither their BIOS or drivers block this."

~~~
jrabone
That's actually good design - the intention being that any clipping can only
be due to software (which you can fix), not underpowered amplification
hardware (which you can't).

~~~
sp332
It's certainly a software issue, but the fault is in the BIOS, not VLC.

------
S_A_P
It would take highly dynamic music to really hear more bits. The quiet
passages is where it makes the difference. Most music is compressed beyond the
need for anything other than 40db of dynamic range.

I barely can tell the difference between high sample rate 8 or 12 bit audio
and 16 bit audio. In some circles I hear people talk about that "gritty" 12
bit sound, but its usually not properly dithered, heavily filtered, 20khz or
less sample rate and generally destroyed on purpose.

Audio is generally the best arena for snake oil since subjectivity is so high.
I cannot believe that Focal Grande Utopia EM loudspeakers(180,000 usd MSRP)
exist in quantity, but I know two people who own a set in their
(comparatively) modest living room. I have heard them in action, they sound
great, but Im not sure that I heard anything in any of the music that I hadnt
heard with good headphones or speakers. At what point is it "good enough"? I
own plenty of recordings that no amount of hifi audiophilia will help.(many
that I recorded myself ;) )

------
pokes
The author is likely using a 16-bit audio card to listen to the result, which
is going to clip the least significant bits off anyway. If you undo the
shifting with sox -v 4096, the quantization noise in the 16-bit version is
audible, even on laptop speakers.

~~~
Breakthrough
This is my only issue with the article - it's alluded that the -v and -b flags
_only_ perform multiplication and bit shifting.

By default, this is __not __true because sox will use a higher bit-count and
perform dithering when converting to a lower bit format
(ref[http://blog.beatunes.com/2014/04/does-24-bit-audio-
matter.ht...](http://blog.beatunes.com/2014/04/does-24-bit-audio-
matter.html?showComment=1397570941495#c7516054401833345657) ) in order to
remove the extra bits while reducing the impact on dynamic range.

In all fairness though, this is the same argument used in the Xiph article (
[http://people.xiph.org/~xiphmont/demo/neil-
young.html](http://people.xiph.org/~xiphmont/demo/neil-young.html) ) in an
attempt to justify the use of 16-bit over 24-bit, the prior allowing for a
comparable dynamic range with the use of appropriate dithering techniques. I
just think it's important to note that the process used with sox is doing more
than simple arithmetic bit-shifts/multiplication.

------
wardb
Weird that the author uses a 'ordinary MP3' as source for the test. Garbage
in, garbage out. Right? Would be a better test with a very high rez audio
file.

~~~
microcolonel
I suspect it's heavily compressed audio. His argument is that you'll have a
hard time hearing it at all, so if anything's going to help you hear it in the
first place, it's garbage in. garbage recordings are the only ones which will
survive this test.

Anyway, nobody here is both qualified and willing at the same time to tell you
what you need to know about this topic. The article referred to, written by
Monty at Xiph, should give you a very good overview of how this works .

[http://people.xiph.org/~xiphmont/demo/neil-
young.html](http://people.xiph.org/~xiphmont/demo/neil-young.html)

~~~
dankoss
> nobody here is both qualified and willing at the same time to tell you what
> you need to know about this topic.

Maybe people who work in professional (not consumer) audio? Who design these
systems for a living?

~~~
microcolonel
Hence "and willing".

------
JazCE
Surely the test is slightly invalidated by converting an MP3 to wav? surely
you would take a master/studio recording and convert that to wav/whatever

~~~
jrabone
The test is most likely invalid - unless told otherwise SoX adds dither. Try
adding -D, I expect the results will be different.

------
rrrooss
Frequency response and dynamic range are low hanging fruit in high res audio
debates. They are not the only factors in accurate reproduction of an input
wave. What is rarely considered is the temporal resolution of the format.
Which is the formats ability to describe change over time. To plot a graph but
depth is the y axis resolution and sample rate the x axis. Again I'm talking
to temporal resolution. Transient response is directly effect by this. The
link below shows the advantages of hi res audio formats inside the human
hearing range. So if accuracy is of concern high res formats DO hold more
information about the original wave form. If file size is of concern hi res is
not applicable.

[http://www.eirec.com/DPimages/digisqwvtest.jpg](http://www.eirec.com/DPimages/digisqwvtest.jpg)
This is an example if transient response of different sampling rates.

~~~
dankoss
I tried to find the context for that image but there wasn't anything on the
site. The image is most certainly incorrect -- or more likely, looking at the
output of the analog electronics after the DAC.

Sampling theory says that a perfect square wave can be represented at any
frequency below Nyquist. That doesn't mean that the codec or the analog
electronics are capable of responding instantly at those frequencies, but that
has nothing to do with the fact that a 1kHz square wave can be perfectly
sampled with a 44.1kHz sample rate. The image is simply incorrect.

Transient response in the real world is generally limited only by the acoustic
transducer response of the system, because everything else has the ability to
respond much faster than audio rates. With Pono, this means that the earbuds
or headphones you use with the player will have a greater affect on the
transient response than the electronics inside.

~~~
rrrooss
This image is from RME a sound card maufaturer. And the image is of the analog
output after dac. So still in the electrical domain. Short comings of a
playback system has what to do with the accuracy of a recoding medium? There
are SO many factors that stop and audio signal reproduction from being
perfect. But the signal domain is the easiest to make better. Transducers that
have to fight the law of physics for accuracy are the obvious weak point. But
as with anything rubish in = rubbish out. Before you even get to the weakest
point of an audio system the transducer (the speaker) transistors and
amplification have their own short comings the accuracy of an amplifiers
transient response is measured by its slew rate. A square wave is impossible
for a speaker cone to reproduce in theory the rising and falling edge is
describing an instantaneous movement. A speaker cone can't do this, inertia
says no. And this is called distortion. As every part of the system introduces
its own distortion the accuracy gets less and less. To accept distortion at
the signal level in the persuit of accuracy is counterintuive. But if your
system can't take advantage of this extra accuracy, go ahead and use a lesser
format, no sense in the extra file size. If you want a reference quality
signal, use the high res formats. To be clear here I'm not saying I want to
listen to digitally encoded square waves, a square wave is a tool for showing
the accuracy of input=output. A square wave being the hardest analog waveform
to capture and reproduce. I am objecting to the idea that hi res formats have
no advantage over what we are used to as consumers. It is the content
distributors that are taking advantage saying that 24bit/96k is more
expensive. The bandwidth and storage does not make this a more premium product
you should be paying for the album not the format. But why limit choice
because the masses can't see the benefits. If you buy the album you as the
consumer can choose the format. I'm not sure if all this arguing stands up to
pick whatever format suits your needs. This argument should be if I bought the
album why can I choose the format most applicable to my playback
needs/desires. And if your concern is that company's charge more for the high
res, that refeclts poorly of the distribution company. If you think it's a
waste of space just download the lesser format. If I had a choice I'd download
it as it left the studio/mastering house.

~~~
dankoss
Not sure what point you're trying to make here, but I was responding to your
post implying that 192kHz sampling rate improves transient response. It
doesn't improve it at all. While there is sampling "distortion" by way of
quantization error, this error is inaudible because we're sampling at 24 bits,
beyond the dynamic range of human hearing. High sample rates actually REDUCE
audio quality because they sample ultrasonics that will interact within the
audible range and potentially damage speakers or pick up EMI interference.

It has been proven over and over via ABX testing that high res formats are
completely indistinguishable from a 24b/44.1kHz master. And further studies
have proven most audio engineers can't distinguish between lossless audio and
320kbps MP3. That's what the Xiph link elsewhere in the thread shows.

------
PavlovsCat
re: "It's useless because most people can't hear the difference."

Sure, but when sampling and then processing things, you can consider it
"headroom". Just like it's not hard to edit a photo in a few steps that the
limitations of 8 bits per pixel begin to show. Though I suspect sampling
frequency is a lot more important there, 24bit can't possibly be "too much".

Yes, I understand that this is not a good argument for consumer products to be
more expensive and need more storage, just so musicians of the future, or
pirate musicians of today, can have a better time. So I think the best plan is
to encourage everybody to become hobby musicians, then it would be an easy
sell.

------
ohwp
Newborns can hear a SNR of about 120dB which is around 21-bit. So 24-bit seems
useless for playing back audio.

But I always wonder how much of the sound we hear by feeling (skin, hair
vibrations). Because maybe I can't hear 1Hz, but I can very well feel it.

~~~
valdiorn
You can definitely "experience" the sub-audio range, but you don't need 24
bits to do that.

I have very sensitive hearing, I hear things most others cannot. I can detect
192kbps vs 320kbps mp3 encoding. I cannot have any switch-mode AC/DC
transformers in my bedroom because the switching noise actually keeps me awake
(I charge my phone in the kitchen, most people think I'm mad when I complain
about the "noise" :)

Still, I've never been able to detect frequencies above 20Khz or hear the
difference between 16 and 24 bit. I think anyone claiming to is full of crap.

~~~
TheLoneWolfling
My laptop whines when I scroll down image-heavy pages. Thus far, I've had a
grand total of one other person comment on it.

------
raverbashing
At the last year's IFA ([http://b2b.ifa-
berlin.com/en/Home.html](http://b2b.ifa-berlin.com/en/Home.html)) there was a
demonstration of a 24-bit audio player with corresponding high quality audio
files and headphones. The results, even though it was a single test, were
_very good_. Of course the issue is, the whole chain has to be 24-bit, if
you're going to pump your 128kbps MP3 there the difference is probably
negligible (not to mention the 1-bit DACs commonly used, that's one of the
elephants in the room)

About the MP3 files, yes, the difference is audible. But of course you need
good headphones.

------
rrrooss
There has been a lot of talk about quantative aspects of but little about the
qualitative. What does a 18khz square wave look like wher recorded and
reproduced in 44.1 kHz. How is phase affecte?. Phase of an audio signal
reaching the ears helps you perceive distance and position. How many samples
are used to represent the frequencies at the higher end of the audible range.
What is the quantization error difference between 16 and 24. And if the author
can't hear the difference between 4bit audio and 12bit audio I question what
he/she is listening on. The aliasing would be HUGE.

~~~
jrabone
An 18 kHz square wave sampled at 44 kHz looks like an 18 kHz sine wave,
everything after the fundamental frequency is well outside the Nyquist limit
and will have been thrown away by anti-aliasing filters. And furthermore you
couldn't hear it even if it wasn't. Fourier decomposition of a square wave
gives the sum of odd multiples of the fundamental, the next frequency is 3 x
18 kHz = 54 kHz.

~~~
rrrooss
Okay now actually look at the response on Matlab. As you get closer to the
nyquist freq there are less samples to describe the wave. And while you get a
bastadised 18k signal it's a far throw from what you put in. Anf while most
are crying your splitting hairs the frequency response and dynamic range
arguments pale in comparison to making the exact wave you put into the encoder
come out. On the right system with the right recording the tiniest nuance in a
room reverb helps trick your brain into believing the sound as actually
happening. And this is not important to all listeners. But people saying that
it makes no difference and is not important as a format to release music in
are thinking only of their current needs and experiences. If you have actually
heard a good recording on a good system in a good room you will know what I'm
talking about. If listening experience has been laptop speakers, headphones
and your mum and dads mini system then I totally agree any more than 320kbs
mp3s are overkill. But to say that high res formats gave no place in consumer
land is to show your lack of understanding of different people and different
needs. In the age of iTunes you surely you buy the album and download it in
whatever format u see fit. If u think audiophiles are wackos go get ya 320's.
I would like to get it as I came fr the studio. The best it can be is as it
came from the studio.

~~~
jrabone
First off, forget about MP3; I'm not arguing for a lossy standard.

You can't get an 18 kHz square wave out of a system with 44 kHz sampling. You
need at least 1 harmonic before it'll even LOOK square, and that requires a
frequency response out to 54 kHz, ie. a sampling frequency of 108 kHz. You
CERTAINLY won't find one in a reverb tail, even assuming you had a generator
for one in the first place (you might JUST get one from a cymbal crash, but I
don't think the physics works)

The point being, your source material can't contain an 18 kHz square wave
either since it's been through a studio production system with the same
antialiasing filters.

Since you know nothing about me but seem to be making assumptions anyway,
here's some background. I've worked in broadcast audio; I own studio
recordings in 24 bit / 192 kHz (Linn release of Mozart's Requiem, studio
master series). I also own studio equipment that can actually play it.
Audiophiles are, by and large, cash cows for companies with no scruples.

~~~
rrrooss
Good an audio nerd. I am making the point about the square wave close to the
nyquist to point out the short comings of a format for accurately reproducing
an input. Square waves in the real world are rare but I am arguing for a
format that produces the most accurate representation of the intended signal.
Imagine the situation where i have my guitar cab set up and I have a square
wave (distortion) coming out of it and I far mic it up so as to capture the
room a give a feeling of space. To really feel like your there you would want
the resulting complex wave made up of the 18k direct sound from the cab and
the room response a recording medium that can't do that accurately is second
rate especialy when the formats are out there. And the higher the sample rate
the further from the nyquist that 18k is and the more samples that can be used
to describe the resulting wave an the more convinced my brain is that sound is
real.

~~~
jrabone
...right up to about 20 kHz, whereafter YOU CAN'T HEAR THEM. Hence 44 kHz
sampling.

Seriously, A/B test this, you might be surprised.

Also if you think that's anything like a square wave coming out of a guitar
speaker (or that that is even desirable in the most case), I've got a bridge
to sell you. And yes, I do play.

~~~
rrrooss
[http://www.eirec.com/DPimages/digisqwvtest.jpg](http://www.eirec.com/DPimages/digisqwvtest.jpg)

Okay here is a picture of what I'm trying to explain. And the author of this
picture used a frequency much further inside human hearing range. This is
transient response test I guess. My main argument is for the verbatim capture
of the input wave. It will make the sound at 10k but it isn't the same wave
that went in.

------
dankoss
The methodology might be flawed, but the point stands: We can't use the full
dynamic range of a 24 bit signal for audio. Great converters have 20-21 bits
of range and if we fully use this range, hearing damage will start at maximum
volume levels. If the volume is kept the same between the two files, there is
a 48dB drop in level - not inaudible, but there are very few source materials
that will actually use the extra dynamic range. On top of that due to fletcher
munson curves we will only hear content in the 1-4kHz range at that low volume
level.

~~~
stinos
_Great converters have 20-21 bits of range and if we fully use this range,
hearing damage will start at maximum volume levels_

but that doesn't have anything to do with number of bits? I mean, an audio
player system is basically a digital stream sent into a DA converter
outputting an analog signal which is then fed into an amplifier where the
signal is amplified and then sent into speakers of some sort. No matter if you
have an 8bit audio stream, or a 128bit audio stream, the stages afterwards
play a much larger role in the hearing damage caused. Also not all DA
converters ouput the same level to begin with. Some are -5V to 5V while others
(single power supply) can be 0V to 5V, etc etc.

~~~
dankoss
Hearing damage as a result of loudness; the electronics have little to do with
it. If you set up a 96dB (16 bit) system such that you can hear the lowest bit
toggling, your maximum volume will be 96dB above that. 96dB SPL is loud but
not painful for average listening.

If you set up a 20 bit system (120dB) so that you can hear the lowest bit
toggling, 120dB above that will most certainly cause hearing damage if used
for very long.

144 dB (24 bit) or more SPL will cause near instantaneous damage and pain.

------
hendriks
For the test the quality of the input does not matter. You can conduct the
same test with a beautiful 24-bit recording as source, you would still not be
able to hear much in the end.

And that's the point.

When you print a photo in size 9x13cm, does it matter whether you print it in
10,000dpi or 20,000dpi? Hardly, unless you always run around with a
microscope.

And it does not matter, whether the photo was originally a good picture or
not.

~~~
PavlovsCat
> When you print a photo in size 9x13cm, does it matter whether you print it
> in 10,000dpi or 20,000dpi? Hardly, unless you always run around with a
> microscope.

It can matter when you put it in a scanner though. I tried scanning some old
photos, they look _horrible_ compared to even my puny 8mp camera. Then I tried
scanning a piece of cloth, and was blown away by the detail..

------
rrrooss
More to it than frequency response and dynamic range. The goal should be for
the most accurate reproduction of the input waveform. Here is a pic of
transient response for different sample rates.
[http://www.eirec.com/DPimages/digisqwvtest.jpg](http://www.eirec.com/DPimages/digisqwvtest.jpg)

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danbee
So I converted a few tracks to 8-bit with noise shaped dither. Have a listen
and remember that the noise floor is now 48dB higher (8-bits) than it would be
with 16-bit.

[https://www.dropbox.com/sh/1h4z0quoybo60c0/O1SwT35Nve](https://www.dropbox.com/sh/1h4z0quoybo60c0/O1SwT35Nve)

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sp332
Did anyone else actually try this? I thought the differences were quite
noticeable. I ran it through Foobar 2000's ABX Comparator and I got the right
answer 5 out of 5 times.

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ntoll
"My God! What has sound got to do with music!" ~ Charles E.Ives
([https://en.wikipedia.org/wiki/Charles_Ives](https://en.wikipedia.org/wiki/Charles_Ives))

(and I'm inclined to agree)

