AES Barcelona 2005: Paper Session L


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	AES Barcelona 2005 Paper Session L - Low Bit Rate Audio Coding, Part 1 (Standard, Systems)

Last Updated: 20050418, mei

Monday, May 30, 14:00 — 16:30

Chair: Jürgen Herre, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany

L-1 DTS-HD: Technical Overview of Lossless Mode of Operation—Zoran Fejzo, Delbert Yee, Keith McDowell, Lorr Kramer, DTS - Agoura Hills, CA, USA
DTS-HD is a multichannel audio codec comprised of a backward compatible DTS core, DTS ES, and DTS 96/24 plus new extensions for additional channels; higher constant bit rates (>1.5Mbps); and lossless mode of operation, and LBR (low bit-rate coding) for streaming applications. In addition to the new coding modes and/or coding components, DTS-HD introduces support for multiple audio assets that can create multiple audio presentations within the single encoded stream. In this paper we present a technical overview of DTS-HD in a lossless mode of operation. We outline the organization and main features of the stream. In addition we will describe the codec architecture and discuss its performance.
Convention Paper 6445 (Purchase now)

L-2 An Introduction to the KOZ Scalable Audio Compression Technology—Kevin Short, Ricardo Garcia, Michelle Daniels, John Curley, Mike Glover, Chaoticom Technologies - Andover, MA, USA
An overview of the high quality, full-bandwidth, lowbit-rate, scalable KOZ audio codec technology is presented. This new compression method grew out of developments in the control of chaotic systems that allow for the creation of broad spectral components with only a few bits of information. These elements are combined with a high-resolution analysis of the audio signal that allows the signal to be decomposed into tonal, noise-like, and transient objects. Psychoacoustic principles have been adapted to prioritize and quantize these objects. The reconstructed signal is built up in layers from the prioritized objects, resulting in a scalable format. Metadata and built-in digital rights management are present in the digital filestream. The decoder is a very low-complexity algorithm that is implemented on a wide variety of portable devices such as cell phones in a software-only solution running on fixedpoint processors without DSP support.
Convention Paper 6446 (Purchase now)

L-3 The Reference Model Architecture for MPEG Spatial Audio Coding—Jürgen Herre, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Heiko Purnhagen, Coding Technologies - Stockholm, Sweden; Jeroen Breebaart, Philips Research Laboratories - Eindhoven, The Netherlands; Christof Faller, Agere Systems - Allentown, PA, USA; Sascha Disch, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Kristofer Kjörling, Coding Technologies - Stockholm, Sweden; Erik Schuijers, Philips Applied Technologies - Eindhoven, The Netherlands; J. Hilpert, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; F. Myburg, Philips Applied Technologies - Eindhoven, The Netherlands
Recently, technologies for parametric coding of multichannel audio signals have received wide attention under the name of "Spatial Audio Coding." In contrast to a fully discrete representation of multichannel sound, these techniques allow for a backward compatible transmission at bit rates only slightly higher than common rates currently used for mono/stereo sound. Motivated by this prospect, the MPEG Audio standardization group started a new work item on Spatial Audio Coding. The paper reports on the reference model zero architecture, as it emerged from the MPEG Call for Proposals (CfP) and the subsequent evaluation of the submissions. The architecture combines the strong features of the two submissions to the CfP that were found best in the evaluation process.
Convention Paper 6447 (Purchase now)

L-4 First Investigations on the Use of Manually and Automatically Generated Stereo Downmixes for Spatial Audio Coding—Bastian Schick, Rainer Maillard, Detmold University of Music - Detmold, Germany; Claus-Christian Spenger, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Recently a new generation of coding technologies was introduced, which—different from the discrete transmission of 5.1 material—only transmit a stereo downmix of the multichannel signal together with a compact side information. From this, the decoder generates a multichannel signal with a spatial image similar to the spatial image of the original input signal. The stereo downmix is created from the multichannel signal using a dynamic downmixing algorithm. Such spatial audio coding techniques also give the opportunity to use a manually created downmix as the transmitted downmix signal for the decoder. In this way, optimal artistic properties of the transmitted downmix signal can be ensured. The paper describes first investigations into the use of manual downmixes in this context including an assessment of decoded surround sound using manual and automatic downmixes.
Convention Paper 6448 (Purchase now)

L-5 Improved Forward-Adaptive prediction for MPEG-4 Audio Lossless Coding—Tilman Liebchen, Technical University of Berlin - Berlin, Germany; Yuriy A. Reznik, RealNetworks, Inc. - Seattle, WA, USA
MPEG-4 Audio Lossless Coding (ALS) is a new addition to the suite of MPEG-4 audio coding standards. The ALS codec is based on forward-adaptive linear prediction, which offers remarkable compression even with low predictor orders. Nevertheless, performance can be significantly improved by using higher predictor orders, more effcient quantization and encoding of the predictor coefficients, and adaptive block length switching. The paper describes the basic elements of the ALS codec with a focus on these recent improvements. It also presents the latest developments in the standardization process and describes several important applications of this new lossless audio format in practice.
Convention Paper 6449 (Purchase now)