Mnyb;696441 Wrote:
So yet again cult audio can come up with the unexpected DAC's that does
sound different at different sample rates :D
Especialy since the most best DACs can´t sound differernt to much
anymore when they measure perfect. When there comes along a DAC that
does everthing
First they need to make the notorious AUTOMIX button that you can find
on some of the cheaper mixing consoles work properly.
--
Soulkeeper
Noise Music Silence
Soulkeeper's Profile:
To boot this to life again .
I've sometimes come across DAC's that don't upsample reedbook but still
are capable of 24/96 there is usually some confused audiophile reasoning
behind this . They may have different filters ( or none ) too.
Have no current example in mind ( there is a new brand of
i'm doing my own dither comparison.
ill post how I see fit, though. thanks. :)
--
TheOctavist
VortexboxSBT(stock)Forssell MDAC-2Klein and Hummell 0300D
Sota Sapphire/Lyra KleosBespoke Valve Phono StageMastersound Due
VentiLink Audio K100
If someone likes i can add the sox dithernoise samples with these 50dB
for easy comparison. I am a bit unhappy with the offered original
sample because it has already shaped noise in but it should be ok.
--
Wombat
Transporter (modded) - RG142 - Avantgarde Acoustic based 500VA
monoblocks -
So i ask you again. Can you tell us where you have the samples in the
dithertest you offered over filedropper?
Did you create them yourself, then please tellme how so i can compare.
I think when you drop in samples you should explain a bit.
--
Wombat
Transporter (modded) - RG142 - Avantgarde
Wombat;695617 Wrote:
So i ask you again. Can you tell us where you have the samples in the
dithertest you offered over filedropper?
Did you create them yourself, then please tell me how so i can compare
on my own.
I think when you drop in samples you should explain a bit.
**The test was
TheOctavist;695661 Wrote:
**The test was performed by reducing the level of a 24 bit file by 50
dB, which causes some of the original signal's level to drop below the
-96 dBFS limit of 16 bit resolution, and then converting to 16 bit in
various ways. The resulting files were then raised in
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
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
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
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
--
TheOctavist
VortexboxSBT(stock)Forssell MDAC-2Klein and Hummell 0300D
Sota Sapphire/Lyra KleosBespoke Valve Phono StageMastersound
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
http://www.ethanwiner.com/dither.html
--
TheOctavist
VortexboxSBT(stock)Forssell MDAC-2Klein and Hummell 0300D
Sota Sapphire/Lyra KleosBespoke Valve Phono StageMastersound Due
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TheOctavist's Profile:
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
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
:)
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
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
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
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:
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
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):
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
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
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
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
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
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
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
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
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
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
--
darrenyeats
darrenyeats;695011 Wrote:
LOL!
Sent from my HTC Sensation Z710e using Tapatalk
I'm here all week...
--
Phil Leigh
You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1
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
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
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
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
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
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
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
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
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
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
http://www.audiorecording.me/how-to-install-run-voxengo-r8brain-sample-rate-converter-in-linux.html
http://www.mainly.me.uk/resampling/index.html
--
TheOctavist
VortexboxSBT(stock)Forssell MDAC-2Klein and Hummell 0300D
Sota Sapphire/Lyra KleosBespoke Valve Phono StageMastersound Due
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
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
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
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
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
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
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
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
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
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
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
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
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.
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
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
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
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
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
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
Ever compared the Beatles USB stick 24bit with the same release CD.
They are different.
Dave
--
DaveWr
DaveWr's Profile: http://forums.slimdevices.com/member.php?userid=9331
View this thread:
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.
--
Phil Leigh
You want to see the
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.
--
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
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
--
DaveWr
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
I replying to this
http://forums.slimdevices.com/showpost.php?p=694674postcount=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
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
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
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
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
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
oops, repost. sorry, but still a good paper
--
TheOctavist
VortexboxSBT(stock)Forssell MDAC-2Klein and Hummell 0300D
Sota Sapphire/Lyra KleosBespoke Valve Phono StageMastersound Due
VentiLink Audio K100
TheOctavist's
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
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
Read my Dan Lavry post, that document gives the full proof !!
Nyquist is an absolute!
--
DaveWr
DaveWr's Profile: http://forums.slimdevices.com/member.php?userid=9331
View this thread:
bhaagensen;694440 Wrote:
Its a nice read and he's probably right in his conclusions - however one
of his main poinsts, as I see it, can not be inferred (only) from what
he's writing. In particular, the reconstruction of a discrete signal
into a continuous one c.f. Nyquist. In fact to the
Thinking about the other end of the spectrum, this is one reason for the
sub versus no-sub debate. By nature, ears register higher frequency
distortion products more than the deeper fundamentals the sub is
producing. And keeping distortion low at low frequencies is very
difficult.
Darren
Sent
Its not that I'm questioning Nyquist - but its a *mathamatical* theory
involving the concept of infiniteness - in particular the
reconstruction theorem involves an infinite sum.
If you read the wikipedia page on this stuff, its noteted that for this
reason, any real-world implementation will
Perhaps look at this this way. The circumference of a circle is pi*d -
easy. Very few people know how to compute this *exactly* on a computer
- not so easy :)
(However its easy to approximate within an error of...)
--
bhaagensen
Irrelevant analogy, just because you use an irrational number in a
calculation, which is always approximated.
On that basis sine waves, which most people believe in aren't accurate,
as they require an infinite series...
sin x = x - x3/3! + x5/5! - x7/7! +
--
DaveWr
If it where not it would not work you would not get =20kHz upper limit
the fact that a DAC reproduce 20k from a 44.1 sampled signal proves
that it is actually working closely enough to Nyquist and likewise if
an ADC samples with 44.1k and still gets 20k it is working ? ( the
theoretical fs/2
The analogt may not be aces, but the point was that in regards to the
article it doesn't say much new that hasnt already been said a zillion
times. The final parts of the argument is missing though I'm sure the
engineers know. There must be some (scientific) epsilon-bound on how
little DACs miss
In conclusion i agree with you mnyb and others. Its just that wrt the
end result the rigidness of mathematical sampling theory does (alone)
not carry through all the way.
--
bhaagensen
bhaagensen's Profile:
bhaagensen;694465 Wrote:
The analogt may not be aces, but the point was that in regards to the
article it doesn't say much new that hasnt already been said a zillion
times. The final parts of the argument is missing though I'm sure the
engineers know. There must be some (scientific)
What a good article. I loved his statement in the outro:
The more that pseudoscience goes unchecked in the world at large, the
harder it is for truth to overcome truthiness...
Almost by way of an example (although he doesn't make the link
himself), in footnote 18 he includes a quote from
bhaagensen;694467 Wrote:
In conclusion i agree with you mnyb and others. Its just that wrt the
end result the rigidness of mathematical sampling theory does (alone)
not carry through all the way.
Human hearing does not carry through all the way either. We are also
not infinite.
Audiophiles
Unfortunately, the picture isn't as clear or as simple as he tries to
portray.
The Boston Audio DBT had flaws which are pointed out in many critiques
all over the net.
Among other things, he seems to thing the point of high res recordings
is so we can hear high frequenies (above 20k); that of
firedog;694484 Wrote:
He talks about macro dynamics of music, but not microdynamics.
But what is microdynamics? Does it have a sensible definition, or is it
just one of those woo-words that can mean anything that the user wants
it to? (something like Spiritual Holistic Quantum Dynamics?)
--
firedog;694484 Wrote:
Unfortunately, the picture isn't as clear or as simple as he tries to
portray.
The Boston Audio DBT had flaws which are pointed out in many critiques
all over the net
Among other things, he seems to thing the point of high res recordings
is so we can hear high
bhaagensen;694457 Wrote:
Its not that I'm questioning Nyquist - but its a *mathamatical* theory
involving the concept of infiniteness - in particular the
reconstruction theorem involves an infinite sum.
I also think this is the weakest spot in the argumentation of the
paper.
While in theory
bluegaspode;694492 Wrote:
So what is done in DACs is to approximate the original waveform with as
much sums as possible in a given timeframe.
NO
The DAC generates a stepped ladder function, the reconstruction filter
turns this into the recovered waveform.
In the case of many modern,
bluegaspode;694492 Wrote:
Let's say for proper playback I can do 100calculations (take whatever
number you like) in 1ms.
Based on 100 calculations: can I get a better approximation of the
original waveform if I have 100 samples or 200 samples ?
NO
Assuming the 100 sampling
This is just wrong.
You don't need an infinite sum or anything, in theory you don't even
need a DAC, all you need is a low-pass filter. A DAC is nothing else
than a number of carefully tuned band-pass filters followed up by a
low-pass filter to clean things up.
No sums involved. No infinity
firedog;694484 Wrote:
He talks about macro dynamics of music, but not microdynamics.
Can you explain what this term microdynamic correpondents with from a
technical standpoint of view?
I lately linked to that thread already if you talk about time
resolution:
As promised link to aliasing tests.
http://www.audiocheck.net/audiotests_aliasing.php
My own ungodly soundblaster cheapest possible souncard aliase above
18kHz
And HF sweeps 22kHz down to 12kHz .
http://www.audiocheck.net/audiotests_frequencycheckhigh.php
These are 44.1 kHz files, so here
Wombat;694512 Wrote:
Can you explain what this term microdynamic correpondents with from a
technical standpoint of view?
I lately linked to that thread already if you talk about time
resolution:
http://www.hydrogenaudio.org/forums/index.php?showtopic=91126hl=
You can always ask J.J.
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