v7.0, 20040922, me

Thursday, October 28, 9:00 am – 11:30 am
Session B LOSSY AND LOSSLESS AUDIO CODING, Part 1

Chair: Michael Goodwin, Creative ATC, Scotts Valley, CA, USA

9:00 am
B-1 MPEG-4 Scalable to Lossless Audio Coding—Rongshan Yu, Institute for Infocomm Research, Singapore, Singapore; Ralf Geiger, Fraunhofer Institute for Digital Media Technology IDMT, Ilmenau, Germany; Susanto Rahardja, Institute for Infocomm Research, Singapore, Singapore; Jürgen Herre, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Xiao Lin, Institute for Infocomm Research, Singapore, Singapore; Haibin Huang, Institute for Infocomm Research, Singapore, Singapore
As the latest extension of MPEG-4 audio coding, MPEG-4 lossless audio coding includes a scalable audio coding solution (SLS) that integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable audio coding into a single coder framework while providing backward compatibility to MPEG Advanced Audio Coding (AAC) at the bit-stream level. Despite its abundant functionalities, SLS still achieves a compression performance that is comparable to state-of-the-art non-scalable lossless audio coding algorithms. As a result, SLS provides a universal digital audio format for a variety of application domains including professional audio, Internet music, consumer electronics, broadcasting, and others. This paper presents the structure of SLS and its latest developments during the MPEG standardization process.
Convention Paper 6183

9:30 am
B-2 Improved Transient Pre-Noise Performance of Low Bit Rate Audio Coders Using Time Scaling Synthesis—Brett Crockett, Dolby Laboratories, San Francisco, CA, USA
A new audio coding tool that uses improved time scaling synthesis techniques has been developed, which reduces the duration of pre-noise introduced by low bit-rate audio coding of transient material. When the transient pre-noise reduction processing is used, decoded PCM audio located prior to transient material is processed in the decoder using time scaling synthesis. The synthesized PCM audio is used to remove or reduce the duration of transient pre-noise, improving the perceived quality of low bit-rate audio coded transient material.
Convention Paper 6184

10:00 am
B-3 Subjective Evaluation of MPEG Layer II with Spectral Band Replication—Gilbert Soulodre, Michel Lavoie, Communications Research Centre, Ottawa, Ontario, Canada
Spectral Band Replication (SBR) was developed as a means of enhancing the coding of audio signals. It has been recently proposed to use SBR, integrated within the MPEG Layer II codec, as a possible extension to the EUREKA 147 DAB standard. The goal is to provide an equivalent level of subjective quality at a reduced bit rate. In the present paper formal subjective tests were conducted to evaluate the performance of Layer II+SBR at typical DAB bit rates. The tests included Layer II+SBR codecs operating at 128 and 160 kbps, as well as a standard Layer II codec at 128, 160, and 192 kbps. The subjective tests were conducted using the ITU-R BS.1534 (MUSHRA) methodology.
Convention Paper 6185

10:30 am
B-4 Spatial Audio Coding: Next-Generation Efficient and Compatible Coding of Multichannel Audio—Jürgen Herre, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Christof Faller, Agere Systems, Allentown, PA, USA; Sascha Disch, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Christian Ertel, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Johannes Hilpert, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Andreas Hoelzer, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; K. Linzmeier, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Claus Spenger, Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; P. Kroon, Agere Systems, Allentown, PA, USA
Recently, a new approach in low bit rate coding of stereo and multichannel audio has emerged: Spatial audio coding permits an efficient representation of multichannel audio signals by transmitting a downmix signal along with some compact spatial side information describing the most salient properties of the multichannel sound image. Besides its impressive efficiency allowing multichannel sound at total bit rates of only 64 kbit/s and lower, the approach is also backward compatible to existing transmission systems and thus accommodates a smooth transition toward multichannel audio in the consumer market. The paper gives an overview of the basic concepts and the options provided by spatial audio coding technology. It reports about some recent performance data, first commercial applications and related activities within the ISO/MPEG standardization group.
Convention Paper 6186

11:00 am
B-5 Coding of Spatial Audio Compatible with Different Playback Formats—Christof Faller, Agere Systems, Allentown, PA, USA
Recently, various schemes were proposed for parametric coding of stereo and multichannel audio signals. Binaural Cue Coding (BCC) is such a technique. It represents multichannel audio signals as a single downmixed channel plus a small amount of side information. BCC can be applied to mono and stereo backwards compatible coding of multichannel audio signals. In this paper we propose a general paradigm for BCC with multiple transmission channels and show how this can be applied not only to bridging between mono/stereo and multichannel surround but also to bridging between different multichannel surround formats.
Convention Paper 6187