Newsgroups: comp.compression
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From: mskuhn@faui09.informatik.uni-erlangen.de (Markus Kuhn)
Subject: Audio Signal Compression (references!)
Message-ID: <mskuhn.673269191@faui09>
Keywords: sound compression
Organization: CSD., University of Erlangen, Germany 
Distribution: comp
Date:  3 May 91 11:13:11 GMT
Lines: 71

Some days ago, I reported in this group about a sound compression
method developped at the University of Erlangen, Germany.
I got a lot of requests for literature, so I went to one
of the local scientists (Rolf Kapust). He gave me a recent
publication (unfortunately only in German):

  Karlheinz Brandenburg, Bernhard Grill, Horst Jonuscheit,
  Rolf Kapust, Dieter Seitzer, Tomas Sporer:
  Uebertragung von hochwertigen Tonsignalen mit Datenraten im
  Bereich 64 bis 144 kbits/s,
  Rundfunktechnische Mitteilungen, Jahrgang 33, Heft 5, 1989.

But the article has an English summary:

  Transmission of high quality audio signals with bitrates
  in the range 64 to 144 kbits/s

  Methods enabling bitrate reduction before transmission and/or
  the recording of high quality audio signals up to factor 7
  without loss of sound quality and up to factor 11 with a
  slight loss of quality are presented. More economic transmission
  or recording methods than those currently used are employed
  for this effect. The electrotechnical faculty of the
  University of Erlangen-Nueremberg has implemented some modern
  methods of bitrate reduction based on psychoacoustic considerations,
  which exploit the properties of the human ear. The error signal
  that results obtained with these methods are described. The
  possibilities of realizing these methods by means of digital
  signal processors are particularly emphasized.

The faculty has -- as far as I know -- a patent on this method.
Especially interesting in the above article is a table comparing 
several compression techniques:

Name     bits/sample     sampling     data     comments
         transmitted       rate       rate
                          [kHz]     [kbits/s]

CD          16             44.1       705.6    just a reference
NICAM       10.1           32         324      simple method
LC-ATC       3             48         144      1 chip solution
OCF          2.5           48         120      quality better than CD!
             2             44.1        88.2    no difference to CD!
             1.45          44.1        64      slight differences to CD

The article contains 15 references. Some of them are:

Zelinski, R.; Noll, P.: Adaptive transform coding of speech
  signals, IEEE Trans. on Acoustics, Speech and Signal Processing,
  Vol. ASSP-25 (1977), p. 299--309.

Brandenburg, K.: OCF - A new coding algorithm for high quality
  sound signals. Proc. 1987 Int. Conf on Acoustics, Speech and
  Signal Processing (ICASSP), p. 141--144.

Brandenburg, K.; Kapust, R. et al.: Fast signal processor encodes
  48kHz/16bit audio into 3bit in real time. Proc. from
  ICASSP, New York 1988.

I am not an expert in this field. If you are seriously interested
in this topic, you should contact these people. I don't know
whether they have an email address. There is no PD source code
available.

Have fun ...

Markus

--
Markus Kuhn, Computer Science student -- University of Erlangen, Germany
E-mail: G=Markus;S=Kuhn;OU1=rrze;OU2=cnve;P=uni-erlangen;A=dbp;C=de
