Newsgroups: comp.compression Path: utzoo!utgpu!news-server.csri.toronto.edu!rpi!zaphod.mps.ohio-state.edu!caen!ox.com!math.fu-berlin.de!fauern!faui43.informatik.uni-erlangen.de!faui09!mskuhn From: mskuhn@faui09.informatik.uni-erlangen.de (Markus Kuhn) Subject: Audio Signal Compression (references!) Message-ID: 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