Randy Yates wrote in message news:QmAyb.377482$Fm2.393962@attbi_s04...
Some Pioneer receivers
run at 88.2 kHz, for example, and current Lexicons run at 96 kHz.
They often use A/Ds at high sampling rates, so analog sources are
digitized at high rates and are processed that way.

Sampling at a high rate in no way obligates the rest of the data path
to operate at that rate. Again, you're making some assertions I don't
believe are true.

Believe what you want. Lexicon prepros operate up to 96 kHz
internally, as confirmed in conversations with Lexicon engineers,
including David Griesinger, and in this Q and A done for the release
of the MC-12:

http://www.smr-home-theatre.org/Lexicon/mc12/qa1.html

"SMR: Lexicon owners are familiar with the outstanding performance
afforded by Logic 7, clearly the most popular of all the available
processing modes. I believe it has been updated and refined yet
further, is that correct?

"Andy Clark: Yes. The MC-12 uses four 32-bit Analog Devices SHARC® DSP
engines, which provide enormous processing power. We have taken full
advantage of this and have re-written the Logic 7 algorithms for 96kHz
internal processing."

CDs are dealt,
usually, in their native rates.

This would make no sense. If the remainder of the data path you speak
of above is operating at 88.2, then it would make sense to upsample
the CDs to 88.2.

The internal architecture of at least the Lexicon, if not other
prepros, operate at multiple sampling rates, and have done so since
the DC-1 introduced in the mid 90s. This is a well-known fact, since
some of its processing modes cannot process above certain sampling
rates. For example, in the DC-1/DC-2/MC-1 architecture, Panorama and
the ambience synthesis modes (Church, Concert Hall, etc.) could not
process 48 kHz sampling rates, and could deal only with 44.1 kHz. In
the MC-12/MC-8 architecture, DTS Neo:6 cannot go above 48 kHz. If
everything were upsampled to 96 kHz, then clearly Neo:6 could not work
at all. Further, since the MC-12/8 can accept 96 kHz digital inputs,
it's trivial to check that there is no decimation and upsampling
happening around the Neo:6 code.

Your statement was that high sample rates are here. I did, and still
do, challenge that remark, if by "here" you mean in widespread market
use. Sure, a small percentage has DVD-A or SACD. (I myself have purchased
a DVD-A - and was abysmally disappointed.) There will always be a small
part of the market buying the most expensive products available. That
wasn't my point, and I don't think it was yours either.

We will just agree to disagree on this. That the internal processing
modes of receivers is already running at 88.1 or higher is a good
enough condition for "here" for me, and that these receivers also have
digital links for DVD-A and SACD with supporting players available
makes it more solid for me.

Frankly, I hope these (SACD and DVD-A) formats fail. CDs are more than
adequate for audio reproduction in any venue barring perhaps a laboratory,
and creating a profuse array of formats does nothing but confuse consumers
and dissipate resources.

They may provide multi-channel music, but we've already got high quality
stereo music. It's called "CD." These formats provide *no* (zero) practical
advantage in music quality over a CD.

Technically speaking, it is incontrovertible that stereo is far
inferior to multichannel. I don't even know how that can be an issue
for discussion, but we can get into it if you like. You had also
originally said: "CDs are more than adequate for audio reproduction
in any venue barring perhaps a laboratory". Tell me how a two-channel
speaker array creates a lateral moving soundwave. Tell me how a
two-channel speaker array gets rid of comb-filtering effects of
phantom imaging. Two channel has been barely adequate for audio
reproduction, and this has been known since the early 30s.

Yes, they will. It is easy to design and implement a half-band filter
using polyphase filtering techniques with fraction-saving or even
noise-shaping that will perform extremely well.

Consider silicon resources vs. latency vs. precision. The choice is
not so easy to make, especially since your typical audio company isn't
going to spin their own ASIC, much less have the technical know-how to
even know such choices exist.

Irrelevent. Generic asynchronous sample rate conversion
is a far, far more complex task than a simple half-band lowpass
filter interpolation (resampling) filter. You're comparing
apples to oranges.

If you had lots of resources to throw at the problem, I agree,
however, most companies don't. This is proven by the vast majority of
companies who use upsampling in their products using the Cirrus
Crystal ASRC part, which makes this relevant. Looking at the specs
published in data sheets for DACs, the fact that they print only THD+N
proves to me that the designers perhaps don't know or care that much
about the human hearing system. And that in turn leads me to view
choices and tradeoffs they make with suspicion.

A mastering engineer can easily screw anything up, even DVD-A and SACD, if
they're not careful or don't know what they're doing. You'll never overcome
ignorance with more technology - only with education.

I agree, but that doesn't change the fact that hi-res audio does
ameliorate many engineering sins, and if that's a way we can get
better sounding music, then so be it.

--Andre