From: j...@research.att.com (jj, curmudgeon and all-around grouch)
Subject: Re: Audio Alchemy's EDR*S Processing Impressions
Date: 1996/06/24
Message-ID: <4qmee1$30a@eyrie.graphics.cornell.edu>#1/1
X-Deja-AN: 161863865
sender: j...@Graphics.Cornell.edu
references: <4q69bj$fb3@tolstoy.lerc.nasa.gov> <4q9e7u$egm@tolstoy.lerc.nasa.gov> <4q9r1k$nob@agate.berkeley.edu>
organization: NJ State Home for Bewildered Terminals
newsgroups: rec.audio.high-end


In article <4q9r1k$...@agate.berkeley.edu> my...@hpfcla.fc.hp.com (Bob
 Myers) writes:

> To get back to the original point, there is simply no way you can pack
> 20 bits of honest-to-good useful information into 16-bit samples
> without doing some form of compression which results in an average
> LOSS of information. 

Whoops. Gotta pick at a nit here, sorry.

If you have a full 20 bits of real, non-redundant information, you're
right.

But that's not what most audio signals look like. Even the
least-redundant audio signal I've ever measured (gets out Akansu and
Smith again), has about -30 dB give or take, spectral flatness
measure, indicating that a purely rate-distortion coding method can
remove about 5 bits on average from that signal, WITHOUT CHANGING THE
VALUES OF THE DECODED SIGNAL.

Now, these two statements are not contradictory, what that -30dB
spectral flatness measure shows is that the signal, while it might
have 16 or 20 bits input, has redundancy (i.e. each sample can be
predicted from the past samples) that can be extracted via well-known
methods and utilized for coding gain.  (That gain at a 64 sample
interval, to be clear.)

In other words, the amount of "information" in a 20 bit signal is at
MOST 20 bits/sample.  It's possible to have much much less.

> The result may or may not be better than what you'd hear having gone
> straight from the analog signal to 16 bits in the first place.

Well, if the signal is typical of audio signals, it's likely to be
possible to do a 1:1 20 bit encode into 16 bits for nearly all samples
of the signal, and very likely all samples given the averages observed
on most audio signals. (A signal that could not be compressed in this
way would be white noise.  I'm not sure we care all that much, but I
must be clear, compressing "white noise" is an oxymoron.)

> I suppose the one advantage I can see in trying to pack 20 into 16
> is that in effect, you would be doing some dithering which could be
> controlled a little better than otherwise.

For most, if not all audio signals, you could probably pack the full
information in the 20 bit signal into 16 bits, in a real process,
that restores the same numbers at the output.  While it's certainly
possible to make a signal for which this wouldn't work, it's not very common
in "real music" signals.

-- 
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