Path: utzoo!utgpu!news-server.csri.toronto.edu!cs.utexas.edu!sdd.hp.com!spool.mu.edu!uunet!math.fu-berlin.de!fauern!faui43.informatik.uni-erlangen.de!faui09!mskuhn From: mskuhn@faui09.informatik.uni-erlangen.de (Markus Kuhn) Newsgroups: comp.compression Subject: Re: Compression of 16-bit sound files. Message-ID: Date: 25 Apr 91 15:12:48 GMT References: <1991Apr21.163913.2249@smsc.sony.com> <1991Apr21.185611.8680@nntp-server.caltech.edu> <1991Apr22.100239.1788@cl.cam.ac.uk> <1991Apr23.221537.21108@cc.tut.fi> Distribution: comp Organization: CSD., University of Erlangen, Germany Lines: 23 Scientists from the University of Erlangen have (as far as I know) developed a lossy sound compression method for voice, music, etc. You can select freely the compression ratio. With 64 kbit/s you get the hifi quality you are used to receive with radio. With 128 kbis/s even experts have _very_ big problems to hear any differences from the original CD data. The algorithm may be implemented in real-time on a DSP and is in discussion for being used in ISDN telephones. The human ear has for each frequency a certain perception level. If a sound is weaker than this level, you have no chance to recognize it. A loud frequency component will cause the perception level to raise for nearby frequencies. The algorithm uses this effect in order to throw away the data you can't hear. If you are interessted in this, I can ask the people hear wether there are any English publications. 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