From: caro@adobe.com (Perry A. Caro)
Newsgroups: rec.audio.high-end
Subject: DCC References (WAS: Audiophile thoughts on DCC?)
Followup-To: rec.audio.high-end
Date: Thu, 15 Oct 1992 18:24:23 GMT
Organization: Adobe Systems Incorporated, Mountain View
	(queueing-rmail) id 173810.3288; Thu, 15 Oct 1992 17:38:10 EDT
Reply-To: adobe!caro@uunet.UU.NET  (Perry A. Caro)
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Originator: tjk@csd4.csd.uwm.edu

Audio (magazine)
September 1991
David Ranada
"NOTEWORTHY: Philips Digital Cassette"
Summary: While details of Philips Precision Adaptive Sub-band Coding
(PASC) are shrouded in trade secrets mystery, an educated guess can be
made of the system based on its close relative, MUSICAM, the digital
radio broadcast compression system.  

Audio
April 1992
David L. Clark
"A/B/Xing DCC"
Summary: Delco Electronics tries double-blind testing of PASC
encoding/decoding vs. straight-through digital music.  Expert listeners
could NOT detect a difference.


There's also an article on the mechanics of the system: format of the
tape, construction of the head, etc., but I don't have the date of that
issue handy.  I'll followup to this posting if I can find it.  I got
these issues out of my local community library, so give yours a try.
----------------------------

About PASC: Assuming the system is similar to MUSICAM (and Philips
engineers reviewing the Audio article said it wasn't far from the
truth) PASC appears to be quite clever.  The basic strategy starts with
18-bits/sample at 1 of 3 sampling rates (48,44.1,32 kHz).  Split the
signal into 32 sub-bands with 24-bits/sample, each being a constant 750
Hz in frequency width.  Meanwhile, take an FFT spectrum of the full
signal.  Do some filtering so you get the "dominant" components.
Calculate masking thresholds (for a given amplitude/frequency
component, there is a threshold function such that components near the
frequency, but below the amplitude threshold, cannot be heard -- e.g.,
a whisper can be heard in a quiet room, but cannot be heard in a room
of rocking party animals).  Now use a psychoacoustic model of the
threshold of hearing to figure out which of the sub-bands is most
important, in terms of preserving fidelity.  It turns out that humans
hear best around 3 kHz, and not surprisingly, most music is
concentrated near that frequency or lower.  A priori artifacts like
that really help in compression algorithms.  Use this information to
allocate the variable number of bits used to encode each sub-band.
Since you will be going from 24-bits to far less (0 to 15), noise is
introduced.  Use an iterative algorithm to minimize the noise (maximize
number of bits used) so that it is below the mask threshold (which may
be higher than the threshold of hearing, or absolute threshold, but
since there is a signal component in the sub-band, it helps to mask higher
level noise).  The rest is formating and scaling (taking advantage of
the fact that across time, signals tend to remain at the same
amplitude, sort of like run-length encoding).

I've glossed over a LOT of details, and of course the source information
is only guesswork to begin with, so caveat emptor.  The upshot is that 
there is sufficient precision in the bits to encode the sounds we can
hear, particularly when high amplitude signals help to mask noise
(thereby reducing the number of bits needed).

About the double-blind testing: I have a lot of respect for David L.
Clark, who does the loudspeaker testing for Audio.  The article
describes how both expert and inexpert listeners had three switches: A
was one form of the source, B was another form of the source, and X
could be either A or B.  By switching between all three, the listener
was supposed to say whether X was A, or B, or couldn't tell.  One of
the signals was pure 18-bit digital from a CD player, the other was the
same 18-bit digital routed through a PASC encoder/decoder.  Appropriate
delays and source material were used.  After 300 trials, listeners did
no better than chance at getting X right.  Though many claimed they did
hear a difference between A and B, those same claimants were no better
than 49% correct at saying whether X was A or B (the percentage would
have to be significantly higher than 50% to indicate a real ability to
detect a difference).

Mr. Clark tried several rigged tests, where he fed white noise to one
channel, and various pathological test signals into the other, trying to
find the boundary conditions of the algorithm.  While he did find some
situations where the output signal was noisier than the input signal, he
could not *hear* the noise -- the noise was below the masking or
absolute threshold.

DCC sounds pretty good!  Please refer to these articles whenever faced
with criticism of DCC being "cold", "limited", "inferior", "harsh",
"tinny", or other voodoo-audio words that will probably be used. :-)

DAT == Betamax?
MD  == 8-track?
DCC == VHS?

Hmm, given playing times and backwards compatability, perhaps this
would be more apt a comparison:

DAT == Hi8?
MD  == Sony's photo disc system, whatchamacallit?
DCC == C-VHS?

Perry
-- 
caro@adobe.com	...!{sun,decwrl}!adobe!caro   Contents: my opinions, no others