🚀 go-pugleaf

"Ulrich Grassberger" <grasso@my-deja.com> wrote in message
news:3b05a36b.30902466@News.CIS.DFN.DE...

Arny wrote:

> >The point is that other than people who've been listening via
> >monitoring systems with lots of high frequency nonlinear
distortion,
> >*nobody* can hear the difference these filters make until the
corner
> >frequency is below 16 KHz.

> >There is BTW a well-documented psychoacoustic reason for this
which
> >I've mentioned on several newsgroups several times and I don't
think
> >there is any informed controversy over. Above about 12 KHz human
> >adults have one critical band. It tends to be captured by the
lowest
> >frequency and strongest sounds in its frequency range. The nature
of
> >musical program material is that there is plenty of content below
16
> >KHz to do that. Since the critical band is dominated by the
content
> >below 16 KHz, the presence or non-presence of content above 16 KHz
is
> >not perceived.


> Well, what if pre-ringing is below 12KHz?

Mr. Grassberger, you'd know that was not how things work in a 44KHz
sampled system if really understood digital audio theory. The ringing
happens at a very specific frequency that is related to the sample
rate, no matter what the frequency of the signal that stimulates the
ringing.

> You�re arguiing about  amplitude, I argue about phase.

One reason I'm not arguing about phase is that I know a little about
the audibility of phase, particularly at high frequencies.

> >"The reason is that: If aliasing will occur not until 48KHz the
> >lowpass filters can be made much simpler and softer. The way in
which
> >signals over 20KHz are cut off in today A/Ds, D/As and resampling
> >filters is brute: Linear-phase filters lead to pre-ringing, an
effect
> >which is absolutely impossible in the analogue world (in the
nature
> >physics that our ears are made for), that is alien stuff. "

> >Pre-ringing seems to happen with musical instrument sounds, not
just
> >test signals. Musical instrument sounds in the last audible octave
> >come from lightly-damped resonators that take a decent number of
> >cycles to build up their resonance, but the still seem to be able
to
> >stimulate pre-ringing, albeit at pretty low levels.

> Yor�re right. I didn�t take this into account, at least not for all
> the time I was working on that paper. Anyway, would you say that
> distortions are unimportant because instruments produce them
anyway?

> >Anybody who wants to can download this file:
> >http://www.pcabx-pro.com/technical/reference/triangle-2_2496.wav
and
> >take a look at it with a 24/96 wave file editor. This file was
> >created using a pair of B&K 4007 measurement mics, an
Audio/Technica
> >48volt phantom power supply, a pair of Benchmark Media MP2 mic pre
(r
> >esponse to 200 KHz), and a CardDeluxe running at 24/96.  There is
> >about 4 milliseconds of background noise before the striking of
steel
> >on steel.  The left channel peaks at 0.00375 seconds from the
> >beginning of the file and the right channel peaks about 0.00390
> >seconds. OK the triangle was not perfectly centered between the
two
> >mics.

> >OK, now lets downsample this file to 1644 with some nice brutal
> >brickwall filtering. That file is called
> >http://www.pcabx-pro.com/technical/reference/triangle-2_1644ds.wav
.
> >If you compare the first 4 milliseconds of the two files you can
see
> >that all sorts pre-ringing is taking place in the downsampled
files.

> >OK so here is the challenge. Prove the  pre-ringing is audible by
> >just listening to the two files, no peeking!

> >I don't think anybody is going to walk way with the prize on this
> >one.

> I didn�t try to *prove* that pre-ringing of digital audio is
> perceptible. That paper is an opinion, and it fits into the
generally
> rebellic style of my site.

Looks to me Mr. Grassberger that we've caught you in a classic
example of the following:

"Yeah, he (Mr. Grassberger) seems to mess up some things some times.
If the results are strange he (Mr. Grassberger) doesn�t bother to
investigate further, to investigate if HE HIMSELF (Mr. Grassberger)
might have made a mistake..."


> >The reason is clear to anybody who understands backwards
> >temporal masking. Here's a reference:
> > http://www.umiacs.umd.edu/~desin/Speech1/node11.html

> Esin Darici Haritaoglu says that signal one will mask signal two in
> the same "critical band", but nothing about the perceptible quality
> (f.e. stereo direction) of signal one.

Let's not run off on a tangent about stereo. You're web site is in
trouble bad enough if we just stick to mono.

Time to shift gears Mr. Grassberger. There are two kinds of masking:
spectral and temporal. Please notice the phrase "backwards TEMPORAL
masking" and then notice the X-coordinate in the figure I referenced
at  http://www.umiacs.umd.edu/~desin/Speech1/node11.html  . Its
milliseconds, obviously time - temporal.

 >"Analogue tape machines are very sensitive for high amplitudes at
> >high frequencies, the dynamic range drops down dead if 20KHz is to
be
> >recorded with -20dB on a cassette recorder."

> >That would be pretty irrelevant to 15 ips or 30 ips master tapes,
> >wouldn't it?

> CDs aren�t recorded on 15ips tape often.

Have it your way. CDs were never specd in terms of their suitability
to be recorded on cassette.

> >"Therefore, when CDs were invented twenty years ago the minimum
> >filter steepness was standardized with 50dB fall in a few
half-tones.
> >(...approximately - I don�t have the papers handy now.)"

> >Yes, brick wall filters have been around for quite a while.
Aliasing
> >is pretty nasty sounding stuff.

> But ultra-high frequency "dirt" is not the same as aliasing.

It's a very common cause. Probably the most common cause.

> >"To show you another point on that field: The HF modulation
frequency
> >in analogue tape machines is higher than 80KHz, simply because it
> >needs to be that high for a smooth upper end."

> >The major benefit of high bias is that it minimizes
intermodulation
> >products that might otherwise fall in the normal audible range,
not
> >smoother high frequency response.

> When I wrote "smooth" I also meant this in a figurative sense,
> "clean". However it shows that a safety margin always is a good
thing,
> both in the digital and analogue world.

Next time you shift gears from the frequency response domain to the
amplitude domain, please try to be a bit more clear.

> >>The issue about electromagnetical wave reflections on audio
cables
> >due to lacking impedance fitting is High-End Voodoo, self-induced
> >("induced", ha ha ha!) imagination, at least with normal cable
> >lengths (and even if it does strike it�s still less audible than
> >pre-ringing), but 96KHz is not.

> >How about a rewrite that makes a little more sense?

> Maybe you do, or some else native English speaker. If someone wants
to
> write an article or correct something, you can, and I�ll give you
big
> credits.

I'm serious. I can't figure out what you meant. I couldn't rewrite it
without the explanation that I'm asking for.

> >>I don�t have any references handy now because as I
> >> stroll thru life I don�t always have in mind that I might need
to
> >> legitimate my opinions to people like you.
>
> >The place to keep your references would be your web pages, no Mr.
> >Grasso?

> The place is in my head. And my site is 15 month old while I�m
somewhat older.

I tried.

Now let's try to correct some more gross technical errors at Mr.
Grassberger's site.

http://grassomusic.de/english/digital.htm  says:

"There is a static noise added to all digital input paths, what is
called dithering. It is made to mask "hard" quantisation distortions
with "soft" noise."

Dither does not mask quantization distortion, it eliminates it. Think
of dither as a means for adding a PWM extension below the lowest bit
of PCM. That's about how it works. A well-dithered system does not
have quantization distortion, period. As the signal level drops in a
well-dithered system, it drops into the noise floor just like what
happens in analog systems.

etc. etc.