Newsgroups: rec.audio
Subject: Re: Compressed audio better than noncompressed?
From: stuartw@ccu1.aukuni.ac.nz (Stuart Woolford)
Date: Mon, 27 Jul 1992 21:53:14 GMT
 <1992Jul24.074842.15614@lugb.latrobe.ed u.au> <22742@oasys.dt.navy.mil>   
 <MONTA.92Jul24121920@image.mit.edu> <1992Jul27.003939.1881@ccu1.aukuni.ac.nz> <23326@alice.att.com>
Organization: University of Auckland, New Zealand.

jj@alice.att.com (jj, curmudgeon for trvth in audio) writes:

>In article <1992Jul27.003939.1881@ccu1.aukuni.ac.nz> stuartw@ccu1.aukuni.ac.nz (Stuart Woolford) writes:
>>monta@image.mit.edu (Peter Monta) writes:
>>
>>>In article <22742@oasys.dt.navy.mil> curt@oasys.dt.navy.mil (Curt Welch) writes:
>>>> My guess is that you can probably do something like 8:1 loss free
>>>> compression on CD's but designing and building the hardware to do this
>>>> in real time is currently too expensive.
>If by "loss-free" you mean "perceptually loss-free", some algorithms
>have been controversially verified by the ISO-MPEG-Audio committee
>to be "transparent" at 2.66 bits/sample, when compression 16 bit samples.
>Nobody I'm aware of (and I do study the field a wee bit) has reported
>transparency <including myself, to the present> below that
>rate, although some algorithms (PAC, ISO-Layer III, for instance,
>there may be others) may provide such performance.

I did acutally mean perceptually-loss free..and should have made that clear..
I have not experienced the ISO-MPEG, so will not comment on it's veracity,
but the theory looks interesting ( but not the way I would go.. ).

>>>If you really mean "loss free", i.e., the original 16-bit data is
>>>reproduced, then this is impossible.  Any audio recording on CD has
>>>an entropy larger than 2 bits/sample.

>>why?? ( no, really, I would like to hear you're reason... )
>Well, because many people have measured it.  If you'd like some
>confirmation, look in Chapter 4 of the book "Advances in Speech
>Signal Processing" by Furui and Sondhi.  Chapter 4, written
>by James Johnston of AT&T Bell Labs, and Karlheinz Brandenburg
>of the University of Erlangen, FRG, is titled "Wideband Coding-
>Perceptual Considerations for Speech and Music".

>On Page 110 and 111, they list some LPC gain statistics and Spectral
>Flatness Measure (SFM) statistics that show a largest SFM
>gain of 45.6 dB at a 512 sample Hann Window.  This, if you
>consider its entropy considerations, means that about half of the
>bits in the signal are actually information-carrying, i.e. you can
>encode this signal in about 8 bits, give or take, and wind up
>with an exact reconstruction.

It seems to me that you are suggesting that at a MAXIMUM (well, you know
what I mean.. ) entropy, remember, I do not look for the constant bit
rates that MPEG ( and MPEG-Audio ?? ) require, as buffers are cheap, and
constant bit rate is not required for locally stored data ( ie:on a CD )
and constant bit rate can KILL compression..

>Other work suggests that for most audio, 5-7 bits/sample is the known
>limit for exact reconstruction, using the most modern techniques.

again, I look for exact AUDIBLE reconstruction.. ie: undetectable by
ANY human ( ie: add something like 10% to top listeners and use that as 
limit.. )

>If one measures the single-sample entropy in a common audio signal,
>perhaps the first 20 seconds of "Tom's Diner" from Solitude
>Standing (Susan Vega), one might find that the single-sample entropy
>is about 8.3 bits/sample.  This, of course, doesn't remove
>intersample redundancy.

then measure the intro to Pink Floyds `sheep` and find that it is about
0.01 bits/sample....

>Good luck using LZW.

personally I prefer Wavelet/VQ followed by a 1MB Markov Chain driving Arith..

>>
>>>> ... As I understand it, both the DCC and MD compression algorithms
>>>> eliminate some sounds based on the human ears ability to here them
>>>> because of masking effects.
>>>> ...
>>>> So, I'm just wondering when people will start claiming that
>>>> compressed audio sounds better than non-compressed becauses
>>>> it has been purified?
>>
>>>Enhancement isn't the goal of these algorithms; they attempt to
>>>minimize the loudness of quantization noise, so to the extent that
>>>the compressed audio is indistinguishable from the non-compressed,
>>>these algorithms are good.  Enhancement/restoration is an interesting
>>>topic, but it's separate from source coding.
>>
>>>Peter Monta   monta@image.mit.edu
>>>MIT Advanced Television Research Program

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stuartw@ccu1.aukuni.ac.nz


			>>>>In VI Where Available<<<<
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Newsgroups: rec.audio