And the fourth group was SB-ADPCM, by NTT and BTRL.
The implementation of the audio part of this broadcasting system was based on a two chips encoder (one for the subband transform, one for the psychoacoustic model designed by the team of G. The simplicity of the corresponding decoder together with the high audio quality of this codec using for the first time a 48 k Hz sampling frequency, a 20 bits/sample input format (the highest available sampling standard in 1991, compatible with the AES/EBU professional digital input studio standard) were the main reasons to later adopt the characteristics of MUSICAM as the basic features for an advanced digital music compression codec.In 1990, Brandenburg became an assistant professor at Erlangen-Nuremberg.While there, he continued to work on music compression with scientists at the Fraunhofer Society's Heinrich Herz Institute (in 1993 he joined the staff of Fraunhofer HHI).As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music. MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg—working as a postdoc at AT&T-Bell Labs with James D.
Johnston ("JJ") of AT&T-Bell Labs—with the Fraunhofer Institute for Integrated Circuits, Erlangen (where he worked with Bernhard Grill and four other researchers – "The Original Six"), with relatively minor contributions from the MP2 branch of psychoacoustic sub-band coders.
The first subgroup for audio was formed by several teams of engineers at CCETT, Matsushita, Philips, AT&T-Bell Labs, Thomson-Brandt, and others. Schroeder at Bell Labs proposed an LPC speech codec, called adaptive predictive coding, that used a psychoacoustic coding algorithm exploiting the masking properties of the human ear. This work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths.