Session Title:	Coding Efficiency
Session Chair:	Marina Bosi, AES Western Region VP, San Francisco, California, USA
Panelists:	Bernd Edler, University of Hannover, Hannover, Germany Takehiro Moriya, NTT Human Interface Labs, Tokyo, Japan, Karlheinz Brandenburg, Fraunhofer Institute for Integrated Circuits, Erlangen, Germany, Eric Scheirer, MIT Media Laboratory, Cambridge, Massachusetts, Deepen Sinha, Lucent Technologies, Bell Laboratories, Murray Hill, New Jersey, USA
Eric Sheirer: "Structured audio" coding schemes provide highly efficient and flexible delivery of certain types of sound over networked and broadband communications media, as well as for virtual environments and video games. We will contrast structured techniques with perceptual coding techniques, highlighting the capabilities of each and their relative advantages and disadvantages for Internet transmission, and discuss synthetic/natural hybrid coding methods. Karlheinz Brandenburg: Compression Efficiency: What are the next steps? Current perceptual coding systems exhibit about the same quality at half the bit-rate compared to the first widely advertised systems of eight years ago. The presentation will give a short overview of: What makes current systems better than their predecessors? Which technologies are in the lab today? What is the progress we might see over the next few years? The focus will be on high frequency resolution perceptual coding systems. Deepen Sinha: Internet multimedia applications present new challenges for audio compression research. Most users still connect to the Internet using a low bandwidth connection (e.g., 28.8 kbps or lower modem), and although emerging networks offer high data throughput, desire for integrated audio/video services and other factors will continue to require vital contribution from audio compression research. Offering high audio quality at the low bit rates offers challenges for all components of the audio coder. Improved supra-thresholding in the psycho-acoustic models, advancements in the signal representation techniques, quantization modules, channel robustness, and embedded coding will likely play an important role. Some of these points will be illustrated in our presentation. Some questions to guide the panel discussion: Has the development of audio coding reached its limits in terms of maximum compression or is there still room for improvement? Are data rates of 64 kb per second per audio channel acceptable for all applications including internet? Which compromise in quality, if any, is the end user willing to accept? How much will the complexity of the coder limit its employment in different platforms? Which approaches are most promising for further reducing the data rates? What are the advantages/disadvantages of using a parametric approach to audio coding? What are the advantages/disadvantages of using a hybrid approach to audio coding? Can we successfully apply speech coding technology to wide-band audio coding? Can we merge functionalities like 3D sound, pitch/time shift, etc., in the basic audio coding scheme? How will scaleability impact the employment of different audio coding schemes? Is interoperability still an issue? What will be the next standard in the audio coding arena?

Biographies:

Marina Bosi, Bernd Edler, Takehiro Moriya, Karlheinz Brandenburg

Marina Bosi Marina Bosi graduated from the National Conservatory of Music in Florence Italy and received a doctorate in physics from the University of Florence, completing her thesis in Paris at the Institut de Recherche et Coordination Acoustique/Musique (IRCAM). During her postdoctorate, dr. Bosi carried out research in sound localization at Stanford University's Center for Computer Research in Music and Acoustics (CCRMA) where she is still a staff member. Dr. Bosi worked for Digidesign developing audio digital signal processing (DSP) technology including dynamic range controller and music analysis/synthesis algorithms. She is currently employed by Dolby Laboratories working on wide-band, low bit rate audio coding. Dr. Bosi is involved with the development of international standards on low bit rate audio coding and is member of ANSI and a US representative in the ISO/IEC WG11 (MPEG), and ITU-R (formerly CCIR) standardization committees. The author of a number of publications on source coding for transmission and storage, her current area of interest is low bit rate coding with applications in music. Dr. Bosi has served the AES San Francisco Section as committee person, vice-chair, and chair. She served as a member of the AES Board of Governors and is currently vice president of the Western Region, USA/Canada. She was co-chair of the 97th AES Convention, for which she received the AES Board of Governors Award, and papers chair of the 101st AES Convention. Dr. Bosi is a member of the technical committee on Audio and Electroacoustics of the IEEE Signal Processing Society and a member of the Acoustical Society of America (ASA). Bernd Edler BERND EDLER received the Diplom (M.S. degree) in Electrical Engineering from the University of Erlangen, Germany in 1985. From 1986 to 1993 he was research assistant at the 'Institut fuer Theoretische Nachrichtentechnik und Informationsverarbeitung' of the University of Hannover, Germany. His main research areas were filter banks and transforms for source coding and he contributed to the development of the filter bank used in the MPEG-1 Layer 3 audio coder. Since 1993 he is staff member of the Systems Technology department at the 'Laboratorium fuer Informationstechnologie' which is a research institution of the University of Hannover. He received the Ph.D. degree in July 1994. His current work focuses on very low bit rate audio coding based on parametric signal representations for MPEG-4. Takehiro Moriya TAKEHIRO MORIYA received B.S. M.S. Ph.D degrees in 1978, 1980, 1989 all from University of Tokyo. In 1980 joined NTT research labs and has been involved in the research for low to medium speech and audio coding. In 1989 he stayed at ATT Bell Labs as an exchange researcher. Now he is a Distinguished Technical Member at NTT Human Interface Labs. Karlheinz Brandenburg KARLHEINZ BRANDENBURG was born in Erlangen, Germany in 1954. He received M.S. (Diplom) degrees in Electrical Engineering in 1980 and in Mathematics in 1982 from Erlangen University. In 1989 he earned his Ph.D. in Electrical Engineering, also from Erlangen University, for work on digital audio coding and perceptual measurement techniques. From 1989 to 1990 he was with AT&T Bell Laboratories in Murray Hill, NJ, USA. He worked on the ASPEC perceptual coding technique and on the definition of the ISO/IEC MPEG/Audio Layer 3 system. In 1990 he returned to Erlangen University to continue the research on audio coding and to teach a course on digital audio technology. Since 1993 he is the head of the Audio/Multimedia department at the Fraunhofer Institute for Integrated Circuits (FhG-IIS). He has presented numerous papers at AES conventions. In 1994 he received the AES Fellowship Award for his work on perceptual audio coding and pscychoacoustics. Dr. Brandenburg is a member of the technical committee on Audio and Electroacoustics of the IEEE Signal Processing Society and the AES. He has been an active member of the ISO MPEG standardization committee since the start in 1988. In MPEG Audio his main topics currently are the MPEG-2 Advanced Audio Coding standard (AAC) and MPEG-4 Audio, where he is the chair of the ad hoc group on MPEG-4 audio core experiments. working on advanced audio coding systems. Dr. Brandenburg has been granted 12 patents and has several more pending.

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