"Arny Krueger" wrote in message
...
"thomh" wrote in message


"Arny Krueger" wrote in message
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"thomh" wrote in message


As you well know Scott, Steve Hoffman is the mastering engineer of
all the AF SACDs and he has said on his website that he uses a
*split feed* coming out of the mastering console which feeds *two*
different A/D converters. So, as Steven Sullivan so succinctly put
it, "A mastering chain that bifurcates at the A/D stage is no
longer the same chain." Any comparisons between the two layers have
to take this into account and it then becomes quite difficult to
pin the differences down to the individual formats.

Where did Steve Hoffman say this?

Here in a discussion concerning the mastering of his Creedence SACDs:

1.
http://www.stevehoffman.tv/forums/sh...&postcount=174

2.
http://www.stevehoffman.tv/forums/sh...&postcount=181

So, his comments about how he mastered this SACD applies only to this one
specific title.


No. AFAIK, this is the usual practice for all the SACDs he's mastered, i.e.
two distinct conversions. No downsampling.

Furhtermore, there could have been any kind of processing you can think of
between the studio and the recorder(s).


Yes, that is true. However, he has stated that the two layers always match
up tonality wise and the difference is only in the resolution where,
according to him, the SACD layer has more. Apparently he has found some old
analog tapes which somehow manages to outperform 4416.

In the same thread, when preaching to his flock, he comes out with
this statement regarding the benefits of hi-res:

http://www.stevehoffman.tv/forums/sh...&postcount=166

IOW: more points = more accuracy.

Does this mean that Steve Hoffman thinks he knows more about digital
resolution than Nyquist and Shannon? Is there a "Steve Hoffman" theorum
related to digital resolution that I don't know about? What issue of the
JAES or the relevant IEEE transactions, or even perhaps the JASA; has
Hoffman's landmark article about digital resolution, and his disproof of
the
theorums of Nyquist and Shannon? ;-)


Shannon and Nyquist has no place on that forum. Nor does double blind
testing, ABX testing or Objectivity vs. Subjectivity.

http://www.stevehoffman.tv/forums/sh...ad.php?t=11234


Here is the whole thread (WARNING: It contains a lot of audiophile
techno-babble and a bunch of ignorant people thanking each other).
BTW, I am the thomh in this thread whose ass was banned for causing
too much trouble and not following the party line. Steve Hoffman
enters the discussions on page 9.

http://www.stevehoffman.tv/forums/sh...5&page=1&pp=20

It turns out that the perceptual coder group at Phillips are pretty
sophisticated about reliable subjective tests. Too bad that this Bruno
Putzeys hasn't taken their work to heart. There is something ironic about
the sound of Bruno Putzeys last name...


Bruno is one of the developers of this piece of equipment:

http://www.grimmaudio.com/ad1grimm.htm

This is a quote from him from a discussion on the Web Mastering Board
http://webbd.nls.net:8080/~mastering/login concerning Dan Lavry's paper on
192k vs. 96k. It is under the section titled 'The Big Controversy' in a
thread called '44.1k vs 48k'.

The second paragraph is pretty interesting in that he obviously thinks DSD
is a waste of time but reckons there is money to be made from it.

QUOTE

Among other things I design discrete AD/DA converters (discrete=consisting
of
standard op amps and logic parts) based on variations of deltasigma.
Recently I've
joined up with three friends to form Grimm Audio (www.grimmaudio.com). The
first product uses a 1-bit modulator operating at 2.8224MHz to put out DSD.

Before anyone cries out "what's a DSD fella doing in this discussion?" I'd
best
explain my position. The existence of DSD as a release format is
unfortunate. My
opinions on the existence of DSD as a production format is such that my
daytime employer
has asked not to proclaim it as openly as I have done earlier. However, SACD
as a
release medium appears to be a commercial success so we should reckon with
it. That's
why I want to provide the best performing tools for people who master
analogue to SACD,
where the only conversion to 1-bit sits at the end of an analogue chain. The
data from
the converter can be spliced together (with only the crossfades reprocessed)
and
otherwise get pressed onto the disc without alteration.

The next logical step for Grimm is also to provide the "best tools"
(converter-wise)
for PCM audio production. Since I expect to be capable of building a
modulator
achieving 120dB over 80kHz, a 192kHz output will be available. In this way I
can
cater for the commercio-emotional needs of the clientele without delivering
a signal that is
degraded compared to what it'd have been had I limited myself to 96kHz
sampling. Decimating
to 192kHz while leaving a noise shaper tail from 40kHz upward would be such
an
instance. Not because of what it sounds like in loop-through mode (=no
effect), but because
of what it'd do to subsequent processing. In that case I'd be compelled to
offer no
higher than 96kHz sampling. Personally I wouldn't waste a tear on it - my
opinions line
up well with Dan's. Of course I could simply cheat like one well-known
high-end
converter supplier, who bluntly quotes a 55kHz bandwidth for 196kHz or DSD
output modes (amidst a total lack of other performance specs).

Digital Filters.
For antialiasing and anti-imaging filters the least bit of sanitary sense
would
dictate the use of filters that properly cut off before nyquist. If everyone
did this,
the move from 48kHz to 96kHz sampling wouldn't have been half as spectacular
as it was.
Also a matter of cleanliness is the use of filters with very low inband
ripple. Inband
ripple translates 1:1 to "echos" at either end of the impulse response. This
is
often tolerated for reasons of group delay (musician's foldback). A reduced
requirement for phase linearity above 20kHz (96kHz assumed) can get you much
lower group delays than allowing pre- and post echos.

"Early roll-off filters"
(Peter Craven calls them Apodising filters) could be seen as the cherry on
top for
those who cringe at the sight of the ringing on the impulse response. When
you realise
that the transmission chain (mic-speaker-ear) is already rolling off quite a
lot,
you'll see that the ringing is already gone.

UNQUOTE

Thom