subject:"\[SlimDevices\: Audiophiles\] 192kHz considered harmful"


http://www.ethanwiner.com/dither.html


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-12 Thread Wombat


TheOctavist;695549 Wrote: 
> Brother, I can't tell any difference with any of them.
> http://www.filedropper.com/dithertest
Yes, i also think it is in practice very unlikely you´ll notice a
difference. 
Thats why i wonder when some audiophiles dig out the big vocabularies
to describe all the audible things tiny changes in the settings of a
resampler do :)

Thanks for the dither samples, now others can have a listen how it
sounds. 
Did you amplify the noise by +48dB or around that? If so i still prefer
the sox Shibata dither curves over the ones in these samples.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Brother, I can't tell any difference with any of them.

I just use the Samplitude 11 inbuilt one..


BTW.

Dither Test.

http://www.filedropper.com/dithertest


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-12 Thread Wombat


TheOctavist;695403 Wrote: 
> this makes no audible difference whatsoever. r8 brain free offers the
> best phase performance with *zero* aliasing..even better than the pro
> version!!
All resampler that have a flat line there have perfect phase response.
How is r8brian best then? 
Avoiding aliasing is common. If you don´t say sox to alias, it doesn´t.
You should better ask yourself if we should allow a controlled amount of
aliasing like Sarcacon for example.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Mynb... if you want to have some more fun, download 'THIS'
(http://www.savioursofsoul.de/Christian/programs/measurement-programs/)

Open your DAW, generate a 1Khz sine wave, then open up Christians tool
and see what is going on. 


It is quite interesting to see the behavior of some of the analog
modeling suites, EQs, and others.!


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Wombat;695127 Wrote: 
> I wouldn´t choose r8brain free, cutting as low as 18kHz is not
> necessary. It even inverts phase.


this makes no audible difference whatsoever.


id bet that you couldn't id them blind.(r8 brain gratis vs pro vs
saracon vs uv22 vs mbit maxx vs POW-R! Hell, I doubt that anyone
could..maybe some of the really badly implemented ones on a certain
type of music.. but doubtful. 

:)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-11 Thread Mnyb


SoX was not that hard to use , the hard part is to choose settings afaik
as SoX is very flexible and probably can be used with anything but sane
settings if you wan't .

Thank you for suggested settings

Preliminary the results are a null ( as expected ) I will convert some
more files too try further 
Haven't picked the exact track to try to ab rigth now I 'm fumbling
with my i pad casually just tapping somewhere in the playlist.
To ab I would mass copy my chosen track and place each one in a folder
of their own then I can have the same file name then I can build a
playlist that I can shuffle and thus hide my own activities to myself.
This will not go down perfectly as I know what I'm testing and already
have an opinion so I'm biased  even if I blind it .

Ideally I should invite a friend and let him/her listen unknowingly of
what's being tested .
But the experience of actually trying on a lot of different tracks put
things in perspective.

I Recommend anyone else to try too :) but use SoX as it is very good,
the paranoia risks are very low as it yields excellent results , I did
a whole zoo of different transcodings but as the diff is very very
small go for 24/96 ( 192 ) to 16/44.1 direct so you can try worst case
.

This is actually very much on topic , I you're unsure on what to think
about the topic just try it for kicks.

This way you can be sure to compare apples with apples commercial disc
with the "same" music in different formats can be compromised.


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-11 Thread Wombat


Mnyb;695242 Wrote: 
> One thing caught my interest here:
> 
> http://src.infinitewave.ca/
> 
> Many brand name name resamplers don't do a spotless job ??
The pictures are nice to watch but as mentioned before you´ll have a
hard time to find even the worst resamplers tested there to sound wrong
:)
Something like the noise down 180dB isn´t exactly needed at all but
most use the stopband so low becaus with todays processing power it
comes easy. Saracon even uses more then these 180dB even but don´t
expect any change especialy when we cut down to 16bit with this
resampling.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-11 Thread Mnyb


Update I used with SoX with 2 suggested comandlines 

The -v option was in my case not really necessary only 1 out 5 files
complained about clipping and it was at most 16 samples so it should be
used on case by case basis.

Unless you *cough* produce loudness war music but then you won't come
here and harass poor old audiophiles about sound-quality *cough* ;)
You would roll in $$$ blowing the ears of the young generation

One thing caught my interest here:

http://src.infinitewave.ca/

Many brand name name resamplers don't do a spotless job ??

No wonder there is some fuzz about it and many engineers get third
party plugins with better performance , A little surprised that the big
names in DAW don't care more . But maybe it's examples from older
versions .
But anyway it's just math why don't do it rigth .

This also again point out SoX as one of the better choices actually
better than what is used professionally in some cases, and that should
cure some hypochondria about the resampling inside LMS .

Some people claims that re-sampling 192 > 96 with LMS/flac/SoX sounds
"bad" 
This is about as BS as it can be.
And constantly whines about that the bad bad naugthy squeezebox only
supports 96kHz :)

I'm going to listen some more on these files with my crappy hifi that
actually only supports 96k, (192k would be re-sampled at the input) ,
the designer thought it was a total waste of power to run all the DRC
and surround modes and digital xovers at 192k .


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Some one have been kind enough

http://src.infinitewave.ca/

Yea I don't hear past 18k anymoore and absolute phase is debated if it
is really audioble on music signals and if it is it the effect is also
gear dependent .

(The 18k to 20k rollof and phase inversion of r8brain free )

Over to use SoX (now it is installed on my main computer )

I urge any other audiophile to try kind of nerd fun and put things into
perpective...


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-10 Thread TheOctavist


I use R8 brain Pro, but to be honest, r8 brain free is more than good
enough.



I have some 24 bit files of guitar, voice, and percussion, and the key
rattles. 

recorded them today.

ill be glad to post the original files(untouched) as well as a
dithered/resampled to 16/44.1 using whatever algorithm you like for
your examination/experimenting..


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-10 Thread Wombat


Mnyb;695182 Wrote: 
> As you described some sound engineers may choose a resampling algorithm
> that is not transparent but give some desired coloration in the process
> ?
I think you misunderstood. There should be no coloration with linear
filters.
I only describe how the perceived noise of the dithershapes sound. This
doesn´t change the sound above, only shifts the "perceived" loudness of
noise further down depending on the taste and audibility.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Wombat;695169 Wrote: 
> I did indeed some testing :) 
> I ended up with this setting (using it with frontah):
> 
> sox.exe input.wav -b 16 output.wav rate -b 92 -a -v 44100 dither -a -f
> low-shibata
> 
> I give up audio content from above 20286Hz with this. 
> With this setting aliasing happens but only above 20800Hz at a low
> level because the response is already pretty down there my testfiles
> show. Since my Transporter cuts from 20600 @44.1kHz it has no chance to
> be a problem.
> One advantage of this early filter is there is nearly no pre-echo but
> that was no criteria for me. You may have seen the pics i linked
> earlier that show the pre- and post-echoes of different approaches.
> This setting also gives me much less clipping as some ultra-steep
> attempts especialy by other vendors. Since clipping is a real problem
> that i stumbled across all the time this is very welcome.
> The sox dither options are well done also. Sox offers different noise
> shaped forms. I tried all by listening them insane loud and decided for
> the low Shibata. It has similar added noise i hear well balanced between
> some HF noise and lower frequency stuff. The Gesemann for example
> doesn´t fit me at all. 
> Really strong high-pitched noise i hear with that.
> You may wonder why iZotope is so hyped. The noise it adds is very much
> low-frequency and sounds most comfortable at first but i don´t like
> it.
> There are much stronger shaped noise-shapes like High-Shibata that
> really seem to allow even lower noise-floors. But with these i am a bit
> cautious. The only time i ever was able to abx a downsampled 24/96 file
> was with some strong shaped dither. The thread is on Hydrogenaudio
> somewhere. I didn´t have the originals to do my own tests and used the
> given samples so this may not be exatly valid.

Interesting , to connect it back to the topic we stablished that a
signal based on 16/48 or 44.1 migth indeed encode all we can hear this
all good .
But is their any way of actually getting there ? " yes " apparently
some resamplers fit this description others don't .
Is there any " gold standard " abx'ed to be inaudible that is well
tested ?

As you described some sound engineers may choose a resampling algorithm
that is not transparent but give some desired coloration in the process
?

This is a slight contradiction why not choose the one algorithm that
gives the least amount of difference especially as the goal is to make
it inaudible , why tweak something your are not supposed to hear ?
Is this some kind of working method from when resampling was not so
well developed .
To reach the objective do a sound engineer chose different approaches
for different projects ? Regarding the resampling .

Starts to wonder if many older releases are compromised by the use of
older less optimal solutions


-- 
Mnyb


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-10 Thread Wombat


Mnyb;695138 Wrote: 
> Will try with SoX as per andy c advice .
> 
> But it is a healthy perspective to muck around with the " defect "
> r8brain resampling too ;)
> 
> Has anyone bothered blind testing algorithms :)
I did indeed some testing :) 
I ended up with this setting (using it with frontah):

sox.exe input.wav -b 16 output.wav rate -b 92 -a -v 44100 dither -a -f
low-shibata

I give up audio content from above 20286Hz with this. 
With this setting aliasing happens but only above 20800Hz at a low
level because the response is already pretty down there my testfiles
show. Since my Transporter cuts from 20600 @44.1kHz it has no chance to
be a problem.
One advantage of this early filter is there is nearly no pre-echo but
that was no criteria for me. You may have seen the pics i linked
earlier that show the pre- and post-echoes of different approaches.
This setting also gives me much less clipping as some ultra-steep
attempts especialy by other vendors. Since clipping is a real problem
that i stumbled across all the time this is very welcome.
The sox dither options are well done also. Sox offers different noise
shaped forms. I tried all by listening them insane loud and decided for
the low Shibata. It has similar added noise i hear well balanced between
some HF noise and lower frequency stuff. The Gesemann for example
doesn´t fit me at all. Really strong high-pitched noise i hear with
that.
You may wonder why iZotope is so hyped. The noise it adds is very much
low-frequency and sounds most comfortable at first but i don´t like
it.
There are much stronger shaped noise-shapes like High-Shibata that
really seem to allow even lower noise-floors. But with these i am a bit
cautious. The only time i ever was able to abx a downsampled 24/96 file
was with some strong shaped dither. The thread is on Hydrogenaudio
somewhere. I didn´t have the originals to do my own tests and used the
given samples so this may not be exatly valid.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Wombat;695127 Wrote: 
> I wouldn´t choose r8brain free, cutting as low as 18kHz is not
> necessary. It even inverts phase.
> 
> This test is pretty outdated. Sox uses the much more refined rate
> effect meanwhile. It is superior to the polyphase effect in several
> ways.
> 
> With sox you can vary the result to pretty much anything you like. The
> different parameters don´t leave anything to be desired.

Will try with SoX as per andy c advice .

But it is a healthy perspective to muck around with the " defect "
r8brain resampling too ;)

Has anyone bothered blind testing algorithms :)

I get on it with SoX to for even more perspective .

Right now I have a set of 5 songs ( originally digitally recorded 24/96
no 40 yo old stuff ) in 24/96 24/48 16/48 and 16/44.1 will redo the lot
with SoX  . Plan is to pick one in one resolution (most likely 16/44.1)
and make a 20 song pl with that and the original and shuffle.

If one scale with sox would the level difference be possible to hear ?


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-10 Thread Wombat


TheOctavist;695079 Wrote: 
> http://www.audiorecording.me/how-to-install-run-voxengo-r8brain-sample-rate-converter-in-linux.html
I wouldn´t choose r8brain free, cutting as low as 18kHz is not
necessary. It even inverts phase.
TheOctavist;695079 Wrote: 
> 
> http://www.mainly.me.uk/resampling/index.html
This test is pretty outdated. Sox uses the much more refined rate
effect meanwhile. It is superior to the polyphase effect in several
ways.

With sox you can vary the result to pretty much anything you like. The
different parameters don´t leave anything to be desired.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-10 Thread andy_c


A couple of years ago I tried some experiments with resampling in which
I downsampled a 24/96 file to 16/44.1, then upsampled again back to
24/96. I used the freeware Audio DiffMaker to take the difference
between the "round trip" resampled file and the original. DiffMaker
determines the optimum scale factor and delay to apply to the "test"
file to get the best null between it and the "reference" file.

When doing this with r8brain, I found clicks in the difference file,
even at the highest quality setting. This caused me to look around for
another resampler. I finally found the command-line utility SoX. The
first thing that happened when I tried it on the same input file was
that it gave me a warning about clipped samples in the downsampled
file. SoX has the option of scaling, and I did this so that clipped
samples were eliminated in both the downsampled file and the "round
trip" downsampled->upsampled one. Then when I found the difference
using DiffMaker, the clicks were gone! This caused me to abandon
r8brain.  The only thing I could hear in the difference file was a
slight amount of hiss, and in order to hear that I had to turn up my
system all the way and put my ear right up against the speaker. Great
care should be used when doing this, as DiffMaker has a bug that causes
an audible blip in the difference file at the beginning, even when
subtracting two identical files.

The command line I used with SoX was like the one below:

sox -v 0.983 track-01-01[0]-01-[L-R]-24-96000.wav -b 16 "01 - Box Of
Rain.wav" rate -v 44100 dither

The first -v specifies that what follows is a volume adjustment. I've
scaled the data by 0.983, which was just barely enough to prevent the
clipping warning message. The next argument is the input file name.
Following the file name is "-b 16", which says the output file should
be 16 bits. Next is the output file name. "rate -v" says to do sample
rate conversion with the -v option, which is the highest quality
conversion available with SoX. Following this is the sample rate of the
output file in Hz. The "dither" option specifies that dither with a
triangular PDF will be used.

SoX is amazingly powerful, and the sample rate conversion above just
scratches the surface of what it can do. I'm quite confident that when
a good resampler is used with the appropriate scale factor needed to
eliminate clipped samples, that resampling down to 16/44.1 will be
transparent in all but the most extreme of cases. I haven't tried it
with really crazy types of files like Tom Danley's fireworks recordings
though.


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


TheOctavist;695079 Wrote: 
> http://www.audiorecording.me/how-to-install-run-voxengo-r8brain-sample-rate-converter-in-linux.html
> 
> http://www.mainly.me.uk/resampling/index.html

Thanks a good read, and r8brain is installed I assume I'll use the very
high quality setting...

I'll have a set of files to with soon this tool is very easy to use
everyone can have a go

Pro's and semi pros seems to have opinions that differ a lot .

Can the professional "gear hypochondria" reach audiophile levels in
extreme cases ? as in if my software has x options and I think I hear
differences between all of them and must obsess over all settings ?

A sligth drop in the 20k region will not be heard by me ( >40yo worked
a lot in heavy industry ) this is an effect of some re samplers .
And obviously not at my comfortable listening levels


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


http://www.audiorecording.me/how-to-install-run-voxengo-r8brain-sample-rate-converter-in-linux.html

http://www.mainly.me.uk/resampling/index.html


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Sure there are a lot of dither/resample choices!@
http://src.infinitewave.ca/

for dither..

there is POW-R (type 1, 2 , 3) triangular dither, Mbit Maxx, Saracon,
UV22(apogee) and dozens of others.


as I have a studio, I am happy to record samples of instruments/voice
and the infamous "keys jangling" (a trial by fire) and give you the
24/96 file and the dithered/resampled ones to examine in whatever way
you see fit!


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread Mnyb


TheOctavist;695074 Wrote: 
> I tell you what. ill take a 24/96 file. 
> 
> ill resample/dither to redbook using 3 different algorithms...you tell
> me.
> 
> interested?

I would not be able to tell :) . But that there is a lot of algorithms
to chose is in itself interesting would not the optimal algorithm be
apparent by now ,when digital audio is a mature technology ?

Curius now :) are you implying that some not optimal algorithm actually
can sound "different" .

but what IS commonly done on the stuff we bought ? (DVDA is now dead )
I have to many disc where it is to different.
There may be seens as an oportinity to release different versions to
give more value a more benign reason than I thougth off .

I'm I over cynical to actually expects differing versions exactly for
the reason to thwart comparisons .

Resampling, is anything free for Linux that works really well maybe I
could learn some command line stuff and use SoX that already lives on
my server .


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Mnyb;695073 Wrote: 
> ? was not this a common case for SACD DVDA ? that there actually where
> different masters chosen the reed-book layer was usually what was
> released as CD in previous releases .
> 
> In some cases it can be more convoluted the DVDA/SACD may be a true 5.1
> mix and has no real 2ch track and people using the stereo out's of such
> player are listening to the automated downmix as provided by these
> formats .
> This will not be the same as the real stereo mix, it may be worse as or
> at least not exactly what the producer intended ?
> 
> A question can the re-sampling and dithering make small level
> differences as a side effect ? 
> 
> 0.2 dB would skew a blind test afaik so just switching between layers
> would not do to compare


I tell you what. ill take a 24/96 file. 

ill resample/dither to redbook using 3 different algorithms...you tell
me.

interested?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread Mnyb


TheOctavist;695067 Wrote: 
> High Resolution tracks dont have any special requirements for different
> mastering than the redbook versions
> 
> normally the redbook is just a dithered/resampled bounce of the
> original high res master...
> 
> 
> Ive never seen a case in which there was different mastering. if the
> redbook master is good, then the HD will to..and vice versa.

? was not this a common case for SACD DVDA ? that there actually where
different masters chosen the reed-book layer was usually what was
released as CD in previous releases .

In some cases it can be more convoluted the DVDA/SACD may be a true 5.1
mix and has no real 2ch track and people using the stereo out's of such
player are listening to the automated downmix as provided by these
formats .
This will not be the same as the real stereo mix, it may be worse as or
at least not exactly what the producer intended ?

A question can the re-sampling and dithering make small level
differences as a side effect ? 

0.2 dB would skew a blind test afaik so just switching between layers
would not do to compare


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


High Resolution tracks dont have any special requirements for different
mastering than the redbook versions

normally the redbook is just a dithered/resampled bounce of the
original high res master...


Ive never seen a case in which there was different mastering. if the
redbook master is good, then the HD will to..and vice versa.


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread R Johnson


mlsstl, It appears that you and I have a rather similar perspective on
this...

One of the reasons I bought the Touch was to be able hear and compare
"high resolution" audio downloads in my own system at modest cost. 
I've decided that, for me, the benefit of high resolution is not worth
a significant price premium.

The "perfect seat" at most venues is usually rather expensive. For
instance, I attend the Chicago Symphony quite often.  For the price of
30 concerts in the Gallery, you might get 10 in the Lower Balcony. 
Now, I do like the Lower Balcony seats (which are much closer, and
"brighter" and louder), but I buy the 30 Gallery seats.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread mlsstl


R Johnson;695023 Wrote: 
> Several years ago I compared the sound of an opera DVD to its
> corresponding HD DVD (before Blu-ray won the war).  I could not hear
> differences in the sound of individual singer's arias.  However, I
> could hear that choral passages were better articulated on the better
> sound track. The audio improvements were subtle. The video improvements
> were very evident.

This is just my personal position, but when the differences between two
formats (or pieces of gear) are in the "subtle" territory, such that I
have to "work at it" to hear the differences, I quickly lose interest
in pursuing the issue. This is particularly true if it is: 1) difficult
to decide if "different" is the same as "better"; 2) there is a
significant cost difference involved; or, 3) there is some aspect of
the "better" option that limits its usefulness for the majority of my
listening. 

The latter is particularly true of high-rez recordings. Much of the
music I buy is simply not available in a high-rez format and what is
available is typically priced much higher. The market has a distance to
go before the exclusivity premium fades to a true production and
delivery cost difference.

Some degree of variability in playback quality doesn't bother me. I
don't get the "perfect" seat every time I go to a live concert, but can
find it always enjoyable if I concentrate on the musical performance.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread R Johnson


mlsstl;694489 Wrote: 
> 
> Can one give an example of a sound that gets lost on a CD that would be
> heard on a higher-rez recording? In addition to the Empire Brass (brass
> is always a challenge to record well), I also listened to some Boston
> Camerata last night, an early music choral group. I could clearly pick
> out individual voices from the 30 or more singers, whether massed, lead
> or background singers. What did the CD lose that a high-rez would have
> revealed? Clearly and accurately presenting the voices of 30 people
> singing together would seem a good test for the ability of a recording
> to maintain clarity and not lose articulation. 
> ...
Several years ago I compared the sound of an opera DVD to its
corresponding HD DVD (before Blu-ray won the war).  I could not hear
differences in the sound of individual singer's arias.  However, I
could hear that choral passages were better articulated on the better
sound track. The audio improvements were subtle. The video improvements
were very evident.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread DaveWr


Wombat;695016 Wrote: 
> Imagine a HD version sounds indeed better because it was less
> compressed. The company will argue they can´t offer less compressed
> versions because the CD media is to limited but with 24/192 they can.
> This is nonsense and i hope more and more peole will get it!
> 
> I´d even go so far to call CD versions that obviously don´t sound as
> good as the HD release while they are from the same version as BROKEN
> and a valid reason to send them back.

Looks like there may be case +1


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread Wombat


rgro;694891 Wrote: 
> This certainly may well be a reason to purchase an HD type recording but
> at least we can now be clear that we're purchasing it for the mastering
> skills of the engineers and not for any real advantage given by higher
> sampling rates or bit depths.
Imagine a HD version sounds indeed better because it was less
compressed. The company will argue they can´t offer less compressed
versions because the CD media is to limited but with 24/192 they can.
This is nonsense and i hope more and more peole will get it!

I´d even go so far to call CD versions that obviously don´t sound as
good as the HD release while they are from the same version as BROKEN
and a valid reason to send them back.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread Phil Leigh


darrenyeats;695011 Wrote: 
> LOL!
> 
> Sent from my HTC Sensation Z710e using Tapatalk

I'm here all week...


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You want to see the signal path BEFORE it gets onto a CD/vinyl...it
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread darrenyeats


Phil Leigh;694900 Wrote: 
> I don't think I am or should be "esteemed", there are others here worthy
> of that epithet... I'd be happy with "tolerated" :-)

LOL!

Sent from my HTC Sensation Z710e using Tapatalk


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread adamdea


bluegaspode;694902 Wrote: 
> Any article that explains it more thoroughly how it is done (hopefully
> with easy explanations and a lot of graphs) ?
> 
> From my understanding: to get the 'correct' value/voltage for a certain
> spot in time I need to add a huge amount of sinc-functions (from
> previous samples and future samples). 
> 
> This was all nicely described in the paper you posted
> (http://www.lavryengineering.com/documents/Sampling_Theory.pdf).
> 
> Now on page 18 I can see that there are circuits which can produce half
> of a sinc function.
> Are such circuits put in parallel to handle many samples at the same
> time to retrieve the final value/voltage of a certain spot in time?
> How many samples/sinc functions does a typical DAC use to get a good
> enough approximation of the final voltage that goes to the speaker?
> And what about the left side of the sinc function?
Agreed. I think this is know at the taps on the filter
I think you might find this interesting
I think the 48 Khz filter used in that article was 153 taps. For fairly
obviously reasons the filter where fs = 96 or 192 had fewer taps 
http://www.nanophon.com/audio/antialia.pdf

For reasons of computational efficiency many dacs (including my own I
think) uised and still use half band filters (you only need half the
number of taps for mathematical reasons I might possibly grasp if i
tried really hard) which means that they are only -6dB at nyquist and
are therefore not sampling theorem-compliant. There is no real excuse
for that these days. I have been meaning to quiz bruno putzeys on this
as a follow up to the reply he gave quoted by Teddy ray.  (octavist)
I think we could improve on this in softeware using sox.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread adamdea


DaveWr;694892 Wrote: 
> In current real world DAC reconstruction, the reconstruction filters are
> analogue, so we have an infinite function - it's an analogue filter.  
> 
> Also no respectable DAC uses 20khz brickwall filters, they have low
> order high frequency analogue filters, as the out of band digital
> products are moved to high frequencies by the use of oversampling
> multibit delta sigma technology.
> 
> Dave
Sort of. Oversampling means that a digital filter takes are of the work
around Fs/2 ansd half the oversampling frequency while an analog filter
deals with the remaining issuees around half the oversampling frequency
upwards.

Unless i am very much mistaken the way this is done illustrates the
point

As part of the opversampling process you still have a form of fitler
which still needs to have the effect of filtering between the audible
range and nyquist. Filtering in the digital domain is a bit of a term
of art which is basically interchangeable with reconstruction/sample
rate conversion etc.

If you look up how a FIR filter works you will see that it needs to
recalculate values at the sample points and it "looks" at a certain
number of sampling points. Because it has to look ahead you have an
inevitable latency created by the filter which is greater the steeper
the filter is- precisely because the steeper the filter the more sample
values is has to recalculate) 

I am a bit at the edge of my understadnign so forgive me if the
following is a buit confused. I am sure there are lots of people here
who understadn this better than I do (feel free to correct) 

I believe that the sample values correspond to the zero corssings of
the sinc function (this makes sense because each sample leaves the
vaslue at other sampling points unchanged -after all we knw the value
at the sampling points). When you upsample however you need to caluate
the values between the existing samples so now each known sample value
does effect the new sample values.Strictly the sinc fnction (which is
an infinitely steep filter) will have an infinite number of zero
crossings which means that each old sample will affect each intersample
value but it can be approximated by using a certin number of sample
points - the number being described as the  number of TAPs on the
filter


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread bluegaspode


DaveWr;694892 Wrote: 
> In current real world DAC reconstruction, the reconstruction filters are
> analogue, so we have an infinite function - it's an analogue filter.  

Any article that explains it more thoroughly how it is done (hopefully
with easy explanations and a lot of graphs) ?

>From my understanding: to get the 'correct' value/voltage for a certain
spot in time I need to add a huge amount of sinc-functions (from
previous samples and future samples). 

This was all nicely described in the paper you posted
(http://www.lavryengineering.com/documents/Sampling_Theory.pdf).

Now on page 18 I can see that there are circuits which can produce half
of a sinc function.
Are such circuits put in parallel to handle many samples at the same
time to retrieve the final value/voltage of a certain spot in time?
How many samples/sinc functions does a typical DAC use to get a good
enough approximation of the final voltage that goes to the speaker?
And what about the left side of the sinc function?


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread Phil Leigh


rgro;694891 Wrote: 
> It would seem that, as several members here---in particular, the
> esteemed Phil Leigh---have been saying for quite some time, the
> differences that people claim to hear so clearly are in the
> engineering/production and not in the bits and bytes.  
> 
> This certainly may well be a reason to purchase an HD type recording
> but at least we can now be clear that we're purchasing it for the
> mastering skills of the engineers and not for any real advantage given
> by higher sampling rates or bit depths.
> 
> Thanks, Octavistfascinating read!

I don't think I am or should be "esteemed", there are others here
worthy of that epithet... I'd be happy with "tolerated" :-)


-- 
Phil Leigh

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ain't what you'd call minimal...
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DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's,
ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-09 Thread DaveWr


Here is another quite well explained approach to digital audio (from
Benchmark)


http://www.benchmarkmedia.com/discuss/feedback/newsletter/2010/08/1/unique-evils-digital-audio-and-how-defeat-them


Basically it says 16bit 44.1khz is now well enough implemented for
music distribution, but not processing.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


adamdea;694872 Wrote: 
> As I understand the most frequently quoted imperfection in real world
> digital reconstruction is the requirement of the perfect sinc function.
> This means that every sample affects the signal between every other pair
> of samples so that the reconstruction filter has to be infinitely long.
> Sort of. But in reality the affect of one sample   Value on the signal
> a few dozen samples away tends very close to zero. So close as to get 
> below the thermal noise of the system. Ie - 120dB or so. I expect
> someone here will have the figures as their fingertips but IIRC 100
> taps would be more than enough.
> 
> This is not new: engineers have for ages had a pretty good idea
> of what is practically perfect. Afaik you can get close enough to
> Nyquist in a dac to keep deviations from perfection outside levels
> which should be detectable using this sort of reasoning. 
> Th
> e possible exception might be the problem of having to get 100db of
> attuation at 22.05 kHz and keep flat response to 20 kHz. According to
> Bruno putzeys maybe you need a 4khz transition band.

In current real world DAC reconstruction, the reconstruction filters
are analogue, so we have an infinite function - it's an analogue
filter.  

Also no respectable DAC uses 20khz brickwall filters, they have low
order high frequency analogue filters, as the out of band digital
products are moved to high frequencies by the use of oversampling
multibit delta sigma technology.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread rgro


It would seem that, as several members here---in particular, the
esteemed Phil Leigh---have been saying for quite some time, the
differences that people claim to hear so clearly are in the
engineering/production and not in the bits and bytes.  

This certainly may well be a reason to purchase an HD type recording
but at least we can now be clear that we're purchasing it for the
mastering skills of the engineers and not for any real advantage given
by higher sampling rates or bit depths.

Thanks, Octavistfascinating read!


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Rg

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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread TheOctavist


oops, repost. sorry, but still a good paper


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread TheOctavist


DaveWr;694726 Wrote: 
> The 16 bit v 24 bit issue is usually audible.  Nobody has been debating
> that, there is science - vastly increased signal to noise ratio, even
> when 24bit tends to be 21/22bit in practice.
> 
> The issue is whether 44.1k or 96k or 192k make a tangible difference. 
> So far nobody seems to have reliably shown this.
> 
> This whole issue is confused because people compare CD with 96k 24 bit.
> Changing two things doesn't lead to the proof of which matters.
> 
> Dave

nope.

http://hlloyge.hl.funpic.de/wp-content/uploads/2007/09/audibility-of-a-cd-standard-ada-loop-inserted.pdf


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Archimago


Henry66;694878 Wrote: 
> You can can do some blind testing of detectable noise floor with 16-bits
> here:
> http://www.audiocheck.net/blindtests_index.php  (the dynamic range
> series)
> 
> I got 9/10 for 72dB with headphones. Very annoying test tone, do not
> wish to repeat.

Cool link, will bookmark that one! Got 10/10 for 72dB with my old
Emu0404USB and Sony MDR-V6's...  Didn't bother with 78dB - as you said,
annoying. Might want to give the Sennheiser HD800 a try later tonight
but since they're not closed back cans, might not do as well...


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Henry66


Wombat;694756 Wrote: 
> Ever listened that loud that you can hear the noisefloor of your 16bit
> gear?
> Just create a file with some silence dithered noise-shaped. Play back
> and turn up the volume until you can hear the noise of that file. Now
> play back an average loud sond, then go shopping for new speakers ;)You can 
> can do some blind testing of detectable noise floor with 16-bits
here:
http://www.audiocheck.net/blindtests_index.php  (the dynamic range
series)

I got 9/10 for 72dB with headphones. Very annoying test tone, do not
wish to repeat.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


darrenyeats;694843 Wrote: 
> I take bluegaspode's point (I think!)
> 
> The effectiveness of Nyquist theory in the real world, given that its
> conditions cannot be met perfectly, must be shown by DBT. It isn't
> enough to say Nyquist is all we need.
> 
> I quote Wikipedia (sorry!) on Nyquist:
> --
> The sampling theorem does not say what happens when the conditions and
> procedures are not exactly met, but its proof suggests an analytical
> framework in which the non-ideality can be studied. A designer of a
> system that deals with sampling and reconstruction processes needs a
> thorough understanding of the signal to be sampled, in particular its
> frequency content, the sampling frequency, how the signal is
> reconstructed in terms of interpolation, and the requirement for the
> total reconstruction error, including aliasing, sampling, interpolation
> and other errors. These properties and parameters may need to be
> carefully tuned in order to obtain a useful system.
> --
> I am not saying Wikipedia is right but the question is out there and
> DBTs are also relevant, and not just arguments based solely on
> Nyquist.
> 
> At the very least, more technical explanation should be presented
> beyond Nyquist given that the requirements of Nyquist cannot be met
> perfectly.
> 
> In my view this debate doesn't appear to be about disagreement on
> conclusions but about the logic and arguments used in support.
> Darren
> 
> Sent from my HTC Sensation Z710e using Tapatalk

As I understand the most frequently quoted imperfection in real world
digital reconstruction is the requirement of the perfect sinc function.
This means that every sample affects the signal between every other pair
of samples so that the reconstruction filter has to be infinitely long.
Sort of. But in reality the affect of one sample   Value on the signal
a few dozen samples away tends very close to zero. So close as to get 
below the thermal noise of the system. Ie - 120dB or so. I expect
someone here will have the figures as their fingertips but IIRC 100
taps would be more than enough.

This is not new: engineers have for ages had a pretty good idea
of what is practically perfect. Afaik you can get close enough to
Nyquist in a dac to keep deviations from perfection outside levels
which should be detectable using this sort of reasoning. 
Th
e possible exception might be the problem of having to get 100db of
attuation at 22.05 kHz and keep flat response to 20 kHz. According to
Bruno putzeys maybe you need a 4khz transition band.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread darrenyeats

I take bluegaspode's point.

The effectiveness of Nyquist theory in the real world, given that its
conditions cannot be met perfectly, must be shown by DBT. It isn't
enough to say Nyquist is all we need.

I quote Wikipedia (sorry!) on Nyquist:
--
The sampling theorem does not say what happens when the conditions and
procedures are not exactly met, but its proof suggests an analytical
framework in which the non-ideality can be studied. A designer of a
system that deals with sampling and reconstruction processes needs a
thorough understanding of the signal to be sampled, in particular its
frequency content, the sampling frequency, how the signal is
reconstructed in terms of interpolation, and the requirement for the
total reconstruction error, including aliasing, sampling, interpolation
and other errors. These properties and parameters may need to be
carefully tuned in order to obtain a useful system.
--
I am not saying Wikipedia is right but the question is out there and
DBTs are also relevant, and not just arguments based solely on Nyquist.

In my view this topic hasn't appeared to be about disagreement on
conclusions but about the logic and arguments used in support.
Darren

Sent from my HTC Sensation Z710e using Tapatalk

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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread darrell


I replying to this
http://forums.slimdevices.com/showpost.php?p=694674&postcount=16 as the
discussion seemed to temporarily jump into a different thread earlier
today!

bluegaspode;694674 Wrote: 
> Well - with some of the denormalized database tables with redundant data
> in our programs it does make our life MUCH easier so that we even don't
> care about the 'waste' of storage space.
> 
> The only difference between database tables and audio I see here is,
> that there seem to have been brave men that even though a provable and
> very correct database normalization theory existed, they still tried
> out if their life could be easier if they dared to just ignore it for
> some tests.
> And they succeeded and proved that there are advantages with doing so.
> 
> With stubbornly adhering to Nyquist theories and not being a little bit
> open minded about the possible usefulness of redundant data how can one
> improve?
> 
> I'm not saying that there will be an improvement with using higher
> sample rates - who knows? But not trying out at all?

What you seem to be saying is that it might be possible to improve on
the (real world) outcome of reconstructing an audio signal from its
digital representation if it is sampled at a frequency way beyond that
required by the Nyquist theory. In support of this you draw an analogy
with the storage of redundant (i.e. unnormalised) data in a database in
order to ease retrieval. 

Others have already explained in detail why all you get when sampling
at a frequency beyond that required by Nyquist is redundant data. A
better database analogy would be IBM's packed decimal format:

We all know that a byte (8 bits) can represent 256 different values.
The packed decimal format takes advantage of the fact that a decimal
digit can have 1 of only 10 values, and therefore stores each decimal
digit in only 4 bits. For example, the decimal value 1234567 is stored
in 4 rather than 7 bytes thus: 12 34 56 7F (the F in the least
significant 4 bits represents the sign - positive in this case). It is
trivial to expand this to the 7 bytes required for display: F1 F2 F3 F4
F5 F6 F7 (in the EBCDIC character set). 

So, we have saved storage space and transmission bandwidth, without
losing any information, and nothing complicated needs to be done to
expand the value for display. 

I hope the analogy holds, but I would say that this is what Nyquist
gives us when we want to store and transmit a bandwidth-limited
waveform.

As far as "not trying it out" is concerned, the original article under
discussion in this post gives links to the double-blind tests which
would seem to confirm what the theory tells us, that there is no
audible difference.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Archimago;694770 Wrote: 
> Great response!
> 
> I respect Barry D and his excellent masters of stuff like ELP, Warren
> Zevon and Bob Marley through the years. But when it comes to hardware
> and opinions on 16/24 bits and sample rates, I really have to wonder.
> 
> A number of months ago I remember him on Steve Hoffman board talking
> about how 24/192 was so much better than 24/96 (I think he even claimed
> this jump was even more significant than 44 --> 96!). Here's a small
> snippet:
> "All that said, I'm glad to see HDTracks introducing the 4x rates. To
> my ears, when they are well done (an important qualifier), 24/176 and
> 24/192 cross a threshold and give us what digital promised from the
> start but is only now starting to deliver - a truly superior medium for
> recording and playback."
> (http://www.stevehoffman.tv/forums/showthread.php?t=238244)
> 
> 
> Then there's this nugget:
> http://www.stevehoffman.tv/forums/showthread.php?t=265346
> 
> Apparently he can hear differences with realtime decoding of FLAC which
> he attributes to jitter (I know I'm digressing here). Oh well...
> 
> Bottom line: Enjoy Barry's work on producing good music with decent
> dynamic range, etc.  His opinions on audio otherwise should be taken
> with large grains of salt / alcohol / hallucinogens.

Agreed. I remember reading his 24/192 comments and wondering whether he
was joking. Bob Katz on the other hand, seems to make sense.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Phil Leigh;694765 Wrote: 
> they do indeed sound different. I'm not sure why though... I wonder if
> the downsampling to 16 wasn't quite done perfectly? Seems odd.

Possibly, I also couldn't honestly say which is to be perceived as
better!

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Archimago


adamdea;694734 Wrote: 
> Deep Deep Deep sigh.
> I have noticed this bloke being cited as an authority, but frankly just
> being a record producer means jack shit.
> 
> As far as i can work out this is a variant on the "distortion of cd at
> -60DB is 100%" nonsense. It is just plain wrong and stupid, and if
> written in the answer to an exam question would simply be marked as
> wrong. 
> 
> It is not "up for debate" or "a matter of perspective" it is just plain
> gloriously wrong, and anyone purporting to know what they are talking
> about who repeats this rubbish should be shunned in polite society, or
> preferably pelted with manure.
> 
> IT IS COMPREHENSIVELY DEALT WITH IN THE ARTICLE LINKED IN THE OP
> (headed the dynamic range of 16 bit)   
> 
> One more time: a correctly dithered 16 bit signal may have broad
> spectrum noise at -93dB, but that doesn't prevent it from being able to
> resolve a signal well below that level. This is because the noise level
> in each portion of the spectrum is much lower than -93Db (even without
> noise shaping). Hence it is demonstrably possible to resolve a tone at
> -120DB in a 16 bit recording.
> 
> 
> So that harmonic information at -36DB below peak? Its still about 80 Db
> louder than the quietest noise you can encode and resolve with 16 bits.
> Even if its -36dB below a quiet noise at -40Db its still only -77dB.
> You have still got 40Db to play with.
> 
> And there is no such thing as "encoded with only 12 bits". if this had
> any meaning you would not be able to record anything with a 1 bit
> stream (DSD).
> 
> So no, Barry Diament has not shown anything.
> 
> and btw I don't an instant believe that he has "shown" that he can tell
> 24 bit files from 16 bit under sensible and meaningful conditions.

Great response!

I respect Barry D and his excellent masters of stuff like ELP, Warren
Zevon and Bob Marley through the years. But when it comes to hardware
and opinions on 16/24 bits and sample rates, I really have to wonder.

A number of months ago I remember him on Steve Hoffman board talking
about how 24/192 was so much better than 24/96 (I think he even claimed
this jump was even more significant than 44 --> 96!). Here's a small
snippet:
"All that said, I'm glad to see HDTracks introducing the 4x rates. To
my ears, when they are well done (an important qualifier), 24/176 and
24/192 cross a threshold and give us what digital promised from the
start but is only now starting to deliver - a truly superior medium for
recording and playback."
(http://www.stevehoffman.tv/forums/showthread.php?t=238244)


Then there's this nugget:
http://www.stevehoffman.tv/forums/showthread.php?t=265346

Apparently he can hear differences with realtime decoding of FLAC which
he attributes to jitter (I know I'm digressing here). Oh well...

Bottom line: Enjoy Barry's work on producing good music with decent
dynamic range, etc.  His opinions on audio otherwise should be taken
with large grains of salt / alcohol / hallucinogens.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Wombat


DaveWr;694762 Wrote: 
> Ever compared the Beatles USB stick 24bit with the same release CD. 
> They are different.
> 
> Dave

Huh? Sure they are different. The sold 24bit version is even louder.
Now make a 16bit version out of the 24bit version yourself. There is
nothing left to be different.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Phil Leigh


DaveWr;694762 Wrote: 
> Ever compared the Beatles USB stick 24bit with the same release CD. 
> They are different.
> 
> Dave

they do indeed sound different. I'm not sure why though... I wonder if
the downsampling to 16 wasn't quite done perfectly? Seems odd.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Ever compared the Beatles USB stick 24bit with the same release CD. 
They are different.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Wombat


DaveWr;694742 Wrote: 
> Agreed although IMHO 24bit is usually an improvement, especially if it
> didn't start life as 16bit!

Ever listened that loud that you can hear the noisefloor of your 16bit
gear?
Just create a file with some silence dithered noise-shaped. Play back
and turn up the volume until you can hear the noise of that fiole. Now
play back an average loud sond, then go shopping for new speakers ;)
This or alike even Daniel Weiss argues agtainmst the need of more bits.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


DaveWr;694726 Wrote: 
> The 16 bit v 24 bit issue is usually audible.  Nobody has been debating
> that
I think you'll find quite a few people around here who would disagree
(myself included).

Nobody on the planet has to my knowledge ever demonstrated an ability
to distinguish 16 bit and 24 bit at normal listening levels where the
only difference is the bit depth.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


ralphpnj;694733 Wrote: 
> Have you looked at the prices for flac downloads in standard redbook
> quality versus 24bit/88.2, 96, 176.4 & 192 kHz? There most definitely
> is a very tangible difference and someone is making lots of money based
> on this difference. Even more money when it turns out (as it sometimes
> does) that the high resolution files are simply upsampled redbook
> files.
> 
> And there are people who say that audio myths are "harmless" :)

Agreed although IMHO 24bit is usually an improvement, especially if it
didn't start life as 16bit!


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


bluegaspode;694728 Wrote: 
> No - quite the opposite. I do believe that Nyquist is right and that in
> theory all this information is enough to recreate the original signal.
> 
> 
> But for some reason I don't believe (ok - lets say I'm at least
> suspicious) if DACs are able to fully adhere to the Nyquist theory and
> are capable of recreating the signal exactly.
> 
> So why is it considered to be better to have 
> a) an oversampling DAC that (due to the Nyquist theorem) can
> approximate the missing information quite correctly (but due to limited
> processing power it will only be approximated)
> 
> b) a DAC that is fed with 192kHz from the very beginning, so wouldn't
> need an approximated oversampling
> 
> Or put it differently:
> Why is a DAC that is given the string "the cat is black" and
> oversamples it to "hheee aattt ss llaaaccckkk" (please
> note the approximation errors in each word that I introduced with my
> oversampling algorithm) considered to produce better/the same results
> as a DAC that is given the more correct "ttthhheee cccaaattt iiisss
> bbblllaaaccckkk" as input from the very beginning?
> 
> Why are people thinking that I don't believe in the Nyquist theorem?
> The theorem has nothing to do with the situation that DACs (for
> whatever reason) are over/upsampling nowadays internally.
> And if that is needed than using the correct input with a higher sample
> rate might yield better results than DACs introducing approximated
> oversampled values?
> 
> 
> 

There are a number of points mixed up here.
1 The real point about the articel cited in the OP is that it argues
that if your downstream equipment is only specified at 20KHz, trying to
reproduce anything above 20Khz is dangerous
- you're just asking it to create spuria above nyquist
- but due to intermodulation distortion is will INEVITABLY create
spuria in the audible range

2 It is thereofre dangerous NOT to filter out everything above the spec
of the downstream equipment.

3 now once you are doing that, what is the point of hi rez file? You
are incurring huge data redundancy as compared with the 16/44 which you
can upsample/oversample to get the benefit of the same digital filters
which you would use with the high res.

Remember you are only approximating the data you are going to filter
out above nyquist. It is true that the higher sample rate material has
a slightly lower noise level in the audible spectrum, but 16 bit is
quite good enough and 24 bits is more than enough already. Using high
sample rates is a very inefficient way of lowering the noise floor in
the audible range for the storage/transmission medium (it is a
different matter at conversion- hence delta sigma dacs) 

17/44 has the same noise floor in the audible range as 16/88 but a much
lower data rate.

4 also don't forget that we can continue to get the benefit of
upsampling well beyond 192k. Childsplay nowadays- but you would be
having insanely huge files if you insisted on using those high sample
rates natively in the stoage transmission mechanism.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


firedog;694717 Wrote: 
> Quote from Barry Diament, record producer who records in hi-res and has
> shown he can pick hi-res versions of his recordings out from standard
> res versions of the same master, reflecting on why 24 bit recordings
> are superior to 16 bit. Note that he isn't talking about hearing 
> frequencies above normal hearing due to the higher k, or saying that
> the information he is talking about doesn't exist on a standard
> recording. Just that useful info that makes instruments sound more real
> and that makes the recorded instruments place in space more clear is
> heard on 24 bit recordings and not as well heard on 16 bit. This is a
> small but useful and audible difference.
> 
> 
> 
> My addition: intrumental harmonics can be 36db lower in level in real
> life.
Deep Deep Deep sigh.
I have noticed this bloke being cited as an authority, but frankly just
being a record producer means jack shit.

As far as i can work out this is a variant on the "distortion of cd at
-60DB is 100%" nonsense. It is just plain wrong and stupid, and if
written in the answer to an exam question would simply be marked as
wrong. 

It is not "up for debate" or "a matter of perspective" it is just plain
gloriously wrong, and anyone purporting to know what they are talking
about who repeats this rubbish should be shunned in polite society, or
preferably pelted with manure.

IT IS COMPREHENSIVELY DEALT WITH IN THE ARTICLE LINKED IN THE OP
(headed the dynamic range of 16 bit)   

One more time: a correctly dithered 16 bit signal may have broad
spectrum noise at -93dB, but that doesn't prevent it from being able to
resolve a signal well below that level. This is because the noise level
in each portion of the spectrum is much lower than -93Db (even without
noise shaping). Hence it is demonstrably possible to resolve a tone at
-120DB in a 16 bit recording.


So that harmonic information at -36DB below peak? Its still about 80 Db
louder than the quietest noise you can encode and resolve with 16 bits.
Even if its -36dB below a quiet noise at -40Db its still only -77dB.
You have still got 40Db to play with.

And there is no such thing as "encoded with only 12 bits". if this had
any meaning you would not be able to record anything with a 1 bit
stream (DSD).

So no, Barry Diament has not shown anything.

and btw I don't an instant believe that he has "shown" that he can tell
24 bit files from 16 bit under sensible and meaningful conditions.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread ralphpnj


DaveWr;694726 Wrote: 
> The issue is whether 44.1k or 96k or 192k make a tangible difference. 
> So far nobody seems to have reliably shown this.

Have you looked at the prices for flac downloads in standard redbook
quality versus 24bit/88.2, 96, 176.4 & 192 kHz? There most definitely
is a very tangible difference and someone is making lots of money based
on this difference. Even more money when it turns out (as it sometimes
does) that the high resolution files are simply upsampled redbook
files.

And there are people who say that audio myths are "harmless" :)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Oversampling DACS are not just simple repetition of previous sample.

The DAC itself is usually a Multibit Delta Sigma DAC typically 4 to 6
bits!

The trick is to take the inputs 16 or 24 bit @ the sampling frequency
and convert them into a 4 to 6 bit value at a very high sampling
frequency.  Since you need more samples they are called oversampling
DACs, but the algorithms used are complex digital filtering techniques
that interpolate the intermediate values.

Most of the DAC manufacturers let the designer select filter types used
for these interpolation filters.  In some cases the product designer
decides to do this whole processing externally to the DAC chip.  The
techniques used do have subtle audio effects.  

This is the same as distortions in amplifiers, loudspeakers etc, some
harmonic some less so.

So why do all this complex stuff.  Well a 4 to 6 bit DAC can be made
very very linear (all steps equal), a 24bit DAC is virtually impossible
with hand built laser trimming techniques.  Secondarily if the
conversion in the delta-sigma DACs are done a very high sample rates,
then the reconstruction filter can be simple, with very accurate
recovery, without the nasty phase and transient effects that sharp high
order filters used to provide.

So yes there is a part in the DAC chain, and the ADC chain that is
based on finite length digital filters.  These are complex, but digital
processing is cheap, and apart from the NOS DAC users many people seem
to recognise that this new breed of DACs has made them good enough,
although obviously not perfect - just like the rest of the audio
recording and playback chain.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread bluegaspode


Phil Leigh;694693 Wrote: 
> 
> 2) You seem to refuse to believe that the (bandwidth limited to below
> Nyquist frequency) information is not accurately captured by sampling
> in accordance with the theory but you have no evidence to base that
> refusal on.
> 
No - quite the opposite. I do believe that Nyquist is right and that in
theory all this information is enough to recreate the original signal. 

But for some reason I don't believe (ok - lets say I'm at least
suspicious) if DACs are able to fully adhere to the Nyquist theory and
are capable of recreating the signal exactly.

So why is it considered to be better to have 
a) an oversampling DAC that (due to the Nyquist theorem) can
approximate the missing information quite correctly (but due to limited
processing power it will only be approximated)

b) a DAC that is fed with 192kHz from the very beginning, so wouldn't
need an approximated oversampling

Or put it differently:
Why is a DAC that is given the string "the cat is black" and
oversamples it to "hheee aattt ss llaaaccckkk" (please
note the approximation errors in each word that I introduced with my
oversampling algorithm) considered to produce better/the same results
as a DAC that is given the more correct "ttthhheee cccaaattt iiisss
bbblllaaaccckkk" as input from the very beginning?

Why are people thinking that I don't believe in the Nyquist theorem?
The theorem has nothing to do with the situation that DACs (for
whatever reason) are over/upsampling nowadays internally.
And if that is needed than using the correct input with a higher sample
rate might yield better results than DACs introducing approximated
oversampled values?

> 
> 3) There are no reliable test results to indicate anyone can hear the
> difference between the same (correctly bandwidth-limited) audio
> information sampled and replayed at 24/44.1 or 24/192. So there is no
> evidence to support your idea that using a higher sampling rate on the
> same bandwidth-limited signal will improve the real-world analogue
> output performance of DAC's. If there were... I'd definitely be trying
> to source true 24/96 or higher recordings just for their better replay
> quality. 

Now this is the right attitude I was missing in all other arguments
before (which were based on pure theory but ignored possible
deficiencies of the hardware).
If someone really tried to record the same signal in 24/192 and 24/44.1
and listeners couldn't tell them apart when played back later in blind
tests ... than it would be some prove that it doesn't matter to the
DACs and/or listeners indeed.

You might have noticed in my previous comments that I'm not arguing
against the whole paper, just this tiny little part about Nyquist which
does not take into consideration the fallacies of nowadays DACs (and
their possible inability to exactly reproduce the waveform they are
supposed to produce according to the Nyquist theorem).
If all this is proved to be below the listeners treshhold - fine, then
I'm happy (and will continue to listen to my 192kB/sec mp3s that only
come through the Receivers jittery digital out ;) )


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


The 16 bit v 24 bit issue is usually audible.  Nobody has been debating
that, there is science - vastly increased signal to noise ratio, even
when 24bit tends to be 21/22bit in practice.

The issue is whether 44.1k or 96k or 192k make a tangible difference. 
So far nobody seems to have reliably shown this.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread firedog


Quote from Barry Diament, record producer who records in hi-res and has
shown he can pick hi-res versions of his recordings out from standard
res versions of the same master, reflecting on why 24 bit recordings
are superior to 16 bit (note that he isn't talking about hearing hi-res
frequencies due the higher k):

> Instrumental harmonics, for example, the parts of the sound that
> differentiate a C played on a guitar from a C played on a piano, or a C
> played on a Baldwin piano from a C played on a Steinway, are much lower
> in level than the fundamental (the C itself).
> 
> Spatial information too - what tells us about the space the players are
> in, the size of the room, its "character", is much lower in level than
> the loudest sounds we hear at a given moment.
> 
> If this information is say, 12 dB lower in level, it will be quantized
> using approximately 2 bits fewer than the total word length (i.e. 14
> bits in a 16-bit encoding, 22 bits in a 24-bit encoding). If it is say,
> 36 dB lower in level, it will be quantized usings approximately 6 bits
> fewer than the total word length (i.e. 10 bits in a 16-bit encoding, 18
> bits in a 24-bit encoding). Some information, such as the end of a
> reverb tail such as in a recording made in a large room, where the
> music ends suddenly, can be well more than that 36 dB lower in level
> than the loudest sounds and will be encoded with correspondingly fewer
> bits.
> 
> This manifests itself in the thinned, bleached and coarsened
> instrumental harmonics in even the best 16-bit recordings, as compared
> to a good 24-bit recording (or of course, the original sound in real
> life). It also manifests itself in the defocusing of the spatial
> information in the 16-bit recording compared to a good 24-bit recording
> (and real life).


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


pippin;694661 Wrote: 
> Err... no.
> Again (why is this so hard to understand???). The discussion is NOT
> about limitations in the sampling process or the reproduction process.
> It's about what of that information you then later need to keep to
> store, transmit and use the data.
I think we are misinterpreting each other's posts. I wasn't trying to
argue with you. I am not suggesting there is any limitation in the
sampling or reconstruction process.

You made a statement along the lines of "All that Nyquist/Shannon says
is ..." and then failed to point out that the signal to be sampled must
be bandwidth limited to below fs/2. So what you stated is not ALL that
Nyquist/Shannon says. I just wanted to pre-empt any attempt by "Nyquist
deniers" to muddy the waters due to this minor ommission.


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Phil Leigh


bluegaspode;694645 Wrote: 
> ...
> I want to listen to music NOW with the best quality possible NOW and
> its of no help if one insists that based on the theory I don't need
> more samples when real world hardware with existing deficiencies might
> still can come closer to the original waveform when fed with more
> samples. Maybe such a discussion is a bit obsolote, now that DACs (seem
> to be) able to do oversampling internally with ease. 
> 
> ...

1) Shannon/Nyquist does NOT require "oversampling" or "upsampling" and
the very idea has no meaning to them. 

"ttthhheee cccaaattt iiisss bbblllaaaccckkk" vs "the cat is black" =
same information, different data oversampling does not alter the
information, only the data.

Oversampling as a technique was invented by DAC manufacturers primarily
to move the recovery artifacts from imperfect analogue filters out of
harms way. Oversampling occurs within the DAC and has no bearing on
storage or transport of information.

2) You seem to refuse to believe that the (bandwidth limited to below
Nyquist frequency) information is not accurately captured by sampling
in accordance with the theory but you have no evidence to base that
refusal on.
On the other hand there is a world of evidence (outside of audio
circles) that sampling theory has proven itself infallible. As a theory
it is stretched much further by the harsh realities of transatlantic or
outer space telecomms than it is by 20-20k hi-fi!.

3) There are no reliable test results to indicate anyone can hear the
difference between the same (correctly bandwidth-limited) audio
information sampled and replayed at 24/44.1 or 24/192. So there is no
evidence to support your idea that using a higher sampling rate on the
same bandwidth-limited signal will improve the real-world analogue
output performance of DAC's. If there were... I'd definitely be trying
to source true 24/96 or higher recordings just for their better replay
quality. 
What I have found is that such recordings or remasterings can sound
great because of improved mastering and that improved sound is still
there if you downsample the files correctly to 44.1. I've tried and
tried and even with my headphones (which are amongst the highest
resolving audio transducers outside of laboratories) on both my own
24/96 and commercial recordings I can't hear ANY difference :-(. If
anyone can reliably claim otherwise, I'd be very interested.

4) If you want to capture, store, transmit and reproduce frequencies
above an arbitary Nyquist limit then the ONLY way to do that is to use
a higher limit through the entire chain. Several respected recording
engineers appear to agree that the 44.1 limit was set a little too low
and that 48 or even 64 would have been a little bit better. However, 96
or 192 or higher are just irrelevant to music reproduction. There's
nothing up there except low level noise and ultrasonic crap... that we
do not want amplified to fry our supertweeters!

YMMV of course :-)


-- 
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ain't what you'd call minimal...
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread Phil Leigh


cliveb;694654 Wrote: 
> Er, no. Traditional oversampling (by factors of 2) is extremely trivial
> - you just stuff zero valued samples in between the existing samples.
> This does not create any extra information - it just alters the
> aliasing artefacts and moves them further up the frequency spectrum.
> Each doubling of the oversampling rate moves the artefacts one octave
> higher. The purpose of oversampling is simply to allow a gentler
> analogue reconstruction filter to be used.
> 
> I'm not so sure about what's now called "upsampling" - where the
> increase in sample rate is not a factor of 2. That does require new
> sample values to be computed. (I'd guess that it's done by first
> oversampling by a large factor of 2, applying a digital reconstruction
> filter, then resampling the result at the desired target rate. Please
> can someone correct me on this). Frankly I cannot see the point of
> upsampling - it acheives nothing of practical value that simple
> oversampling doesn't.

The main benefit (or I should say "aim" or "intent") of ASRC (hardware
upsampling) is to reduce certain types of jitter. It does indeed invent
new sample values.  I think this is a "swings and roundabouts"
situation, personally. Many DAC's use ASRC's AND internal
oversampling!


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread bluegaspode


I'm not questioning Nyquist?

pippin;694659 Wrote: 
> 
> Again. I know you don't want to understand it: but if you need more
> samples, YOU CAN MAKE THEM UP. It's cheaper, it's easier and it's even
> BETTER!
> 

Ok - then where is the prove of that?
Not the prove of that one can make the up (that is proved by Nyquist
and I don't question that at all).
I question the part about the "cheaper", "easier" and "better".


If I was audiophil and mean I would now even through in the argument,
that the less computation a DAC has to do (i.e. less over/upsampling
for whatever reason that is needed internally) the better the produced
audio/ less jitter/less humptydumpty.

Of course such a theory also can only be proved by doing (blind) tests,
but not trying at all because in theory that is not needed is a bit
stubborn in my opinion.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread pippin


cliveb;694649 Wrote: 
> 
> What you have failed to point out is that if you sample at 2*X, and if
> the signal being sampled contains any frequency components greater than
> X, then the result does NOT accurately encode the information up to X -
> it will include aliasing components below X that were not in the
> original sample. This is why the signal needs to be band-limited before
> sampling. I'm sure you understand this and not mentioning it was simply
> an oversight. But I mention it so that you get nit-picked by friend
> rather than foe :-)

Err... no.
Again (why is this so hard to understand???). The discussion is NOT
about limitations in the sampling process or the reproduction process.
It's about what of that information you then later need to keep to
store, transmit and use the data.
Aliasing is adding noise (or reducing the quality of your signal)
during the sampling process. I never did or will claim that any
sampling technology is perfect, none will ever be.
And yes, that first paper as well as your argument indicate that using
a smaller band for sampling actually helps with the quality of the
result but I would not claim that it's impossible to do good recordings
at higher samplerates, too.

But that wasn't my point about Shannon/Nyquist.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread pippin


bluegaspode;694645 Wrote: 
> Come on. It doesn't make sense to adhere to a perfect theory when the
> hardware cannot come close to what the theory claims (as said: I don't
> know if DACs currently come close or not).
> 
It can. All this is about is transmission and storage of data.
And we do have hardware that can do that perfectly, so no problem to be
solved here.
> 
> I want to listen to music NOW with the best quality possible NOW and
> its of no help if one insists that based on the theory I don't need
> more samples when real world hardware with existing deficiencies might
> still can come closer to the original waveform when fed with more
> samples.
> 
Yes. The point is: it can't. Theory or not.

> 
> Maybe such a discussion is a bit obsolote, now that DACs (seem to be)
> able to do oversampling internally with ease. 
> 
Again: The DAC has nothing to do with this. It's all about the bits on
your harddisc only.

Handling more data actually causes a lot of real-world problems that DO
degrade you audio experience: lack of disc space, lack of bandwidth,
higher power consumption, processing requirements, noise, heat and
reduced battery life of components and still we haven't even entered
the DAC.
All just to shovel around redundant data that you could as well create
using a cheap 2ct a piece logic component right before the DAC.

> 
> So the question comes up: what is cheaper to produce: a DAC that
> internally can do oversampling with very high accuracy or a DAC without
> oversampling that is just fed with 192kHz?
> 
No. The question is: is it cheaper to blow up 48 kHz samplerate-file
right in front of your DAC using a shift register or to transport,
store and process it end-to-end.

> 
> Or taking examples from other domains, where theory alone does not
> help:
> 
> Why are there error correction bits on (data) CDs? In theory they are
> not needed as bits are bits. But in reality the hardware has
> deficiencies (i.e. scratches) so we deliberately put extra bits on the
> disc to overcome the deficiencies.
> 
Again: you are mixing information theory and technical implementation.
The CD production process is lossy. If you don't trust your harddrive,
add error correction bits, too. But you would not add three empty
tracks to your CD in the hope of making the chances of reading the
non-empty ones better by some obscure theory.

> The same logic also applies to Nyquist theorem.
> 
No. And you can repeat that as often as you like. The Nyquist theorem
is a mathematical theorem and it holds. Always. No need for fudge-ups.
Fudge up your data storage, transmission, DACs, loudspeakers,
microphones, mixing equipment, whatever, but there is no need to fudge
up your data.

> So my main question remains: are there known deficiencies in DACs that
> are easier to overcome with more samples or is it cheaper to work on
> the deficiences thus just improving the DAC that gets fed the same
> input?
> 
Again. I know you don't want to understand it: but if you need more
samples, YOU CAN MAKE THEM UP. It's cheaper, it's easier and it's even
BETTER!


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


bluegaspode;694645 Wrote: 
> But is it really that easy? Reading the paper (both the one provided
> first, but even more the one posted by DaveWR) I come to the conclusion
> that oversampling in the digital world isn't trivial at all, because I
> need a big enough set of points from sinc-functions from previous and
> future samples to get the actual oversampled value between two provided
> samples.
Er, no. Traditional oversampling (by factors of 2) is extremely trivial
- you just stuff zero valued samples in between the existing samples.
This does not create any extra information - it just alters the
aliasing artefacts and moves them further up the frequency spectrum.
Each doubling of the oversampling rate moves the artefacts one octave
higher. The purpose of oversampling is simply to allow a gentler
analogue reconstruction filter to be used.

I'm not so sure about what's now called "upsampling" - where the
increase in sample rate is not a factor of 2. That does require new
sample values to be computed. (I'd guess that it's done by first
oversampling by a large factor of 2, applying a digital reconstruction
filter, then resampling the result at the desired target rate. Please
can someone correct me on this). Frankly I cannot see the point of
upsampling - it acheives nothing of practical value that simple
oversampling doesn't.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Pippin,

Overall you make good points, but you missed an important issue:

pippin;694598 Wrote: 
> All that Nyquist/Shannon says is: if you have a frequency X, which is
> the maximum frequency you are interested in (here: the highest
> frequency you could probably ever hear), then if you use a sampling
> frequency of 2*X to store your sampled data, then ALL information
> contained in the signals below X will be in included in the information
> you store. There is NO additional information you get by using a higher
> sampling frequency. Nothing. All you get is additional information
> about frequencies ABOVE X but not below, the information on the
> frequencies below X is already there and it's complete.
> 
What you have failed to point out is that if you sample at 2*X, and if
the signal being sampled contains any frequency components greater than
X, then the result does NOT accurately encode the information up to X -
it will include aliasing components below X that were not in the
original sample. This is why the signal needs to be band-limited before
sampling. I'm sure you understand this and not mentioning it was simply
an oversight. But I mention it so that you get nit-picked by friend
rather than foe :-)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-08 Thread bluegaspode


pippin;694627 Wrote: 
> 
> No, that, too, is wrong. The theory is correct. Just because your
> filter is bad doesn't mean that any different data will serve it
> better. Worse. Now you need to do an end-to-end optimization just to
> avoid using perfect information. Doesn't make sense and will certainly
> not work.
> 

Come on. It doesn't make sense to adhere to a perfect theory when the
hardware cannot come close to what the theory claims (as said: I don't
know if DACs currently come close or not).

I want to listen to music NOW with the best quality possible NOW and
its of no help if one insists that based on the theory I don't need
more samples when real world hardware with existing deficiencies might
still can come closer to the original waveform when fed with more
samples. Maybe such a discussion is a bit obsolote, now that DACs (seem
to be) able to do oversampling internally with ease. 

> 
> There is no issue with upsampling 44.1/16 to 12.3GHz@486 bits if that's
> what your DAC needs.

But is it really that easy? Reading the paper (both the one provided
first, but even more the one posted by DaveWR) I come to the conclusion
that oversampling in the digital world isn't trivial at all, because I
need a big enough set of points from sinc-functions from previous and
future samples to get the actual oversampled value between two provided
samples. 
So when talking about good enough oversampling we are in the domain of
buffer sizes/look aheads and computation power and compromises will be
necessary as we cannot do 'correct' oversampling based on
Nyquist/Shannon in the digital world with limited computational power.

So the question comes up: what is cheaper to produce: a DAC that
internally can do oversampling with very high accuracy or a DAC without
oversampling that is just fed with 192kHz?


Or taking examples from other domains, where theory alone does not
help:

Why are there error correction bits on (data) CDs? In theory they are
not needed as bits are bits. But in reality the hardware has
deficiencies (i.e. scratches) so we deliberately put extra bits on the
disc to overcome the deficiencies.

Noone would ever start to argue that these extra bits are a waste,
because in theory with perfect CDs that cannot be harmed by scratches
they are not needed. And also noone tries to build perfect
scratch-resistant CDs now, but just accepts that we use some extra bits
to workaround some deficiences of the hardware.

The same logic also applies to Nyquist theorem. It is NOT enough to
just point to the theorem and then stop all discussions about higher
sample rates (which I think is a flaw in that part of the paper, though
the other parts surely compensate this).

So my main question remains: are there known deficiencies in DACs that
are easier to overcome with more samples or is it cheaper to work on
the deficiences thus just improving the DAC that gets fed the same
input?

Just pointing to Nyquist does not answer such a question.
Always adhering to the perfect theory is too expensive in the real
world, so workarounds dominate here.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread pippin


bluegaspode;694626 Wrote: 
> 
> Let's say the hardware that is used to draw the circle, is only able to
> work with a rough estimation of PI (2 instead of 3.14...). The circle
> that this device would be drawing would be a very bad approximation of
> the original circle and every graphophile lover of circles would
> complain. Now in such a scenario you will draw BETTER circles with 100
> samples of points on the perimeter and even better circles with 1000
> samples of points on the perimeter.
> 
Sorry, but that's nonsense.

We are talking about digital information processing here.
If your hardware that does the drawing of the circle only can use rough
approximations, you need to convert your perfect circle (center plus
radius) to whatever strange approximation your machine needs to draw a
good circle.
All you need for that is a computer that understands pi and how to do a
perfect circle from the center point and the radius (and of course your
machine's limitation).

The information (center point and radius) is still complete. There is
nothing else you will ever need to describe it.

Even worse. What you are postulating assumes that a WORSE
representation of the circle could actually lead to better results
through a process that involves that the CREATOR of the limited
approximation actually has a better understanding of the limitations of
your reproduction machine than that machine itself (because it's
creating a worse approximation of the circle to assist the latter's
reproduction process).

Actually a pretty accurate description of a lot of stuff being done in
audio but still by no means sensible.

> 
> So as long as a DAC (or the reproduction filter) isn't working with
> good enough sinc waveforms the Nyquist theory remains just a nice
> theory.
> 
No, that, too, is wrong. The theory is correct. Just because your
filter is bad doesn't mean that any different data will serve it
better. Worse. Now you need to do an end-to-end optimization just to
avoid using perfect information. Doesn't make sense and will certainly
not work.

> 
> Maybe you all know so much more about DACs than me, that you are not
> questioning the sinc functions in your DAC anymore.
> 
No, we have understood Shannon and so we know it's two completely
separate problems.

Again (that's what I wrote above): if your DAC needs something else
than 44.1/16, it can CREATE IT FROM THAT. There is no issue with
upsampling 44.1/16 to 12.3GHz@486 bits if that's what your DAC needs.
There will not be any information loss. But you gain exactly nothing
from transmitting and storing data in that format.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode


pippin;694598 Wrote: 
> 
> All of this has nothing, really NOTHING to do with the topic at hand.
> It just doesn't matter how good or bad the DAC or the ADC or the
> speaker or the microphone is. All of this has absolutely NOTHING to do
> with the storage format for the music.
> 

You are missing my point.
It's fundamental that I am asking about how DACs work, because
otherwise the Nyquist theorem is of no value in our practical world.

To make my argument clear lets go back to circles: I think there is
some theory that claims that all data you need to draw a perfect circle
is to have it's center point and radius (that would be 2 samples).
Without asking how the circle will be drawn later, it is WRONG to just
conclude that it doesn't help to record more samples (like 100 points
on the perimeter of the circle).

Let's say the hardware that is used to draw the circle, is only able to
work with a rough estimation of PI (2 instead of 3.14...). The circle
that this device would be drawing would be a very bad approximation of
the original circle and every graphophile lover of circles would
complain. Now in such a scenario you will draw BETTER circles with 100
samples of points on the perimeter and even better circles with 1000
samples of points on the perimeter.

So as long as a DAC (or the reproduction filter) isn't working with
good enough sinc waveforms the Nyquist theory remains just a nice
theory.
Maybe you all know so much more about DACs than me, that you are not
questioning the sinc functions in your DAC anymore.

But I am :) ... and maybe also many audiophiles that read that article,
so don't stop at claiming 'Nyquist is enough, there is no more to talk
about'.
If you want to prove that in nowadays world higher samplerates don't
help you cannot stop at Nyquist but you need to go all the way to the
loudspeaker.

> 
> You are exactly right - virtually all modern DACs are multibit
> delta-sigma DACs. These bring another filter that does affect sound
> quality - the interpolation filter. This is used to manufacture samples
> that don't exist in the original sampling.
> > > > 
> > Uhh ohhh. And why is this better than just using on of the original
> > samples if I had the double sample rate? Or are we in DAC snake oil
> > territory already when it comes to creating interpolation filters that
> > invent some new samples?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb


Thanks pippin .

It is not about specific techniques let's assume that it is sota the
best 192k stuff is used during mastering and playback 

I to so tries to make this point if we just leave bithdept depth aside
for a while .

Ponder for a moment that no analog signal recorded had any content
above > 24 kHz

Then a 48k signal and a 192k signal would contain exactly the same
signal it would reconstruct to exactly the same wave form you would not
even forensically be able to tell them apart .

What is said is that a digital signal has the necessary data to
completely describe any signal at fs/2.

this includes the time domain there is a fallacy to believe that the
content between samples are lost that there is an inaccuracy of +_ 1/fs
this is false.

So arguments for higher fs necessary implies that it should be any
merit of playing back ultrasonics .


There may be 1000's of reason for a studio or DAW software to operate
at any level .


As pippin and many others say to get that high quality 44.1 or 48
signal any kind of higher fs recording and processing can be involved
but that's not really the point , on the contrary employing such
methods will probably yield a perfect such signal for all intents and
purposes and actually prove the point that fs at 44.1 encodes all info
in the frequency domain we can hear.
But all beside the point


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread pippin


bluegaspode;694580 Wrote: 
> 
> I obviously never cared about how DA converters work, based on the
> paper I now think of thousands of  of sinc-producing circuits which all
> add up to the final waveform. 
> Probably this is not how it works in practice but this a missing link
> for me now to agree that Nyquist theorem and good circuits is all that
> we need.


We have to do away with a huge misunderstanding here, I believe.
This article is NOT about how a DAC should work, which technology is
best and what kind of limitations practical recording and playing
equipment may have.

All of this has nothing, really NOTHING to do with the topic at hand.
It just doesn't matter how good or bad the DAC or the ADC or the
speaker or the microphone is. All of this has absolutely NOTHING to do
with the storage format for the music.


All that Nyquist/Shannon says is: if you have a frequency X, which is
the maximum frequency you are interested in (here: the highest
frequency you could probably ever hear), then if you use a sampling
frequency of 2*X to store your sampled data, then ALL information
contained in the signals below X will be in included in the information
you store. There is NO additional information you get by using a higher
sampling frequency. Nothing. All you get is additional information
about frequencies ABOVE X but not below, the information on the
frequencies below X is already there and it's complete.


What we have to understand here, is that this has nothing to do with
any imperfections in the recording or playback process, these will of
course be there, but if your recording is crap, i won't get better just
because you STORE more of it and at a higher sample rate. And if your
DAC is distorting, then it will not get any better just because you
throw higher frequencies at it - to the contrary, the article implies
it's getting actually WORSE because of effects letting distortions from
the higher frequencies leak into the lower frequency spectrum.
To be clear: this is NOT "missing information" that was just not
recorded due to the low sample rate, it's DISTORTED information due to
the bad reproduction process in the DAC.


What the article does NOT say is that it doesn't make sense to use
different sample rates or sample sizes for processing. It can make
sense to use something different while processing your data, for
example because of limitations of the technology you use. A good
example is the 24 bit sample size used in the Squeezebox internally.
This makes perfect sense because what the Squeezebox does is it does
digital processing to change the volume. If you stick to 16 bit data,
you would get rounding errors and information losses due to this
processing that you can avoid if you go to 24 bit in processing.
But it does NOT mean, that anything gets better if the data you throw
at it is already 24 bit.

To use a somewhat different analogy: When a bank calculates interest,
it will use 4 additional digits behind the cent (so 1$ is 1.00  for
them). Why? Because if you get, for example, 1% of interest for your
dollar per year and that interest is paid monthly, the monthly interest
you get would amount to 0.00 0833 ct. If you round that, you just get 0
so you would never get any interest which would be plain wrong because
to calculate things right, they will have to pay you 1 ct per year.
HOWEVER, they will never actually PAY you 0.0833 ct because there is no
such thing. and your Dollar doesn't get any different just because you
write it as 1.00  $, it's still exactly the same thing as 1$ and
actually everything behind the last cent digit has no meaning at all
(or it would already be a rounding error).


Likewise, nobody says that there will be no way to invent some fancy
technology that does a more accurate recording of the analog audio
signal at 2 MHz sample rate and this can be superior to a 16 bit 44.1
kHz microphone.
HOWEVER: If you then take the digital output of that hypothetical
processor and down-convert it to 44.1 kHz samplerate audio, then for
all frequencies below 22.05 kHz, there will be NO, not even the
slightest loss of information.
So it doesn't make sense to STORE and TRANSMIT the data at higher
frequencies.

There are a few good arguments for 48 kHz, most notably that since it's
common to use 96kHz or 192 kHz equipment in processing (remember: it can
STILL make a lot of sense to do PROCESSING at higher frequencies,
especially in the digital domain) you get pretty much simplified
up/downsampling logic.
I can't argue about 24 vs. 16 bit, that does indeed depend on the
actual dynamic range you can record and reproduce and I don't know
where technology is here, purely from an information  theory
standpoint, 24 bit word size DOES contain more information than 16 bit
word size, that's different from the sample rate thing.


Now there is a third thing, and that's the "trust your ears" thing.
1. Yes, you should, because in the end it's all that matters
2. Normal people do that but as

Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Soulkeeper;694588 Wrote: 
> AFAIU, delta-sigma DACs are used for PCM decoding. It's the most
> widespread type of audio DAC, at least according to some of what I've
> read.
> 
> Unfortunately the article I linked to went into more detail about ADC
> than DAC.
> 
> After a quick scan of the DSD article on Wikipedia, I get the idea that
> DSD stores audio in a delta-sigma modulated format, while delta-sigma
> DACs convert a PCM signal to a delta-sigma modulated signal as part of
> the DA conversion.

You are exactly right - virtually all modern DACs are multibit
delta-sigma DACs.  These bring another filter that does affect sound
quality - the interpolation filter.  This is used to manufacture
samples that don't exist in the original sampling. This is usually
where designers claim all their specialities.  For example, Linn in
their DS products don't use the DAC interpolation, but their own design
of interpolation and noise shaping digital filters.  Whether this is
different / better is probably a mute point.  I think it is very much a
low level effect.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper


AFAIU, delta-sigma DACs are used for PCM decoding. It's the most
widespread type of audio DAC, at least according to some of what I've
read.

Unfortunately the article I linked to went into more detail about ADC
than DAC.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


Soulkeeper;694585 Wrote: 
> I found 'this'
> (http://skywired.net/blog/2011/05/introducing-the-delta-sigma-modulator/),
> which looks promising. I'll start reading it myself, now.

Those systems are 1 bit A/D and D/A systems.  This is the technology
used by Sony in their DSD techniques as used in SACDs.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper


bluegaspode;694580 Wrote: 
> Is there any more (hopefully easy to understand) information about how
> these reconstruction filters work in practice?

I found 'this'
(http://skywired.net/blog/2011/05/introducing-the-delta-sigma-modulator/),
which looks promising. I'll start reading it myself, now.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful


maggior;694553 Wrote: 
> Just trying to educate myself here...
> 
> Does this mean that there could be audible distortion introduced due to
> ringing in a recording that has clipped waveforms?  Taken to an extreme,
> clipping could start to approximate a square wave.
> 
> BTW, I'm finding this discussion to be very interesting...I'm learning
> a lot.

Clipping itself is an extreme distortion, at medium power levels due
the high frequency content of these waveforms, this can easily destroy
loudspeaker tweeters.

The clipping waveform will usually have some of this high frequency
removed by the anti-alias filter that is the first part of the ADC
(analogue to digital convertor).  Although it will be now more benign,
it is still a distortion.

The ringing issue from square waves is no longer any issue, due to the
way ADCs are designed.  they don't have a very sharp (often 7th order)
'brickwall' filter at 20khz anymore.  This was always the source of
some low level errors, also repeated in the playback DAC chain.  10
years ago it was trendy to have multiple filter choices in your DAC
chain of your CD player, and they did sound slightly different.

As with all things digital speeds have gone up, and the digital guys
made it easier to get more linear ADC / DAC systems by using only a few
bits but at very high oversampled rates to achieve better results to
standard multibit DACs.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode


Phil Leigh;694539 Wrote: 
> I
> What some people are misunderstanding here is that it is the
> reconstruction filter that recovers the analogue signal, not the DAC.
> The DAC simply presents the filter with a set of voltages over time. It
> is within the filter that the sinc function becomes manifest and this is
> indeed an infinite series - the mathematical definition of a filter is a
> continuous function over time. A filter is not a step function!
> To be clear on this, what comes out of the filter IS a mathematically
> perfect sine wave... 

I read through the paper posted before in the meantime
(http://www.lavryengineering.com/documents/Sampling_Theory.pdf )

Is there any more (hopefully easy to understand) information about how
these reconstruction filters work in practice? The paper comes close
around page 18, where it is shown how at least the right half of a sinc
function can be produced by some circuit.
I'm missing the left part of the sinc function because based on the
explanation it is need as well to recreate the original wave-form.

I obviously never cared about how DA converters work, based on the
paper I now think of thousands of  of sinc-producing circuits which all
add up to the final waveform. 
Probably this is not how it works in practice but this a missing link
for me now to agree that Nyquist theorem and good circuits is all that
we need.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper


The article linked in the OP covers bit depth too. 

> The incorrect '96dB' figure ignores the spectral power density of a
> signal. 16 bit audio can go considerably deeper than 96dB, and deeper
> yet with proper use of dither. Handled correctly, the dynamic range of
> 16 bit audio reaches 120dB in practice [10], more than twenty times
> deeper than the 96dB claim.
> 
> That's greater than the difference between a mosquito somewhere in the
> same room and a jackhammer a foot away or the difference between a
> deserted 'soundproof' room and a sound loud enough to cause hearing
> damage in seconds.
> 
> 16 bits is enough to store all we can hear, and will always be enough.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful