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P20 - Low Bit-Rate Audio Coding, Part 2Monday, May 22, 14:00 — 15:20
Chair: Mark Vinton, Dolby Laboratories - San Francisco, CA, USA
P20-1 A Novel Integrated Audio Bandwidth Extension Toolkit (ABET)—Deepen Sinha, ATC Labs - Chatham, NJ, USA; Anibal Ferreira, ATC Labs - Chatham, , NJ, USA, University of Porto, Porto, Portugal; Harinarayanan E. V., ATC Labs - Chatham, NJ, USA
Bandwidth Extension has emerged as an important tool for the satisfactory performance of low bit-rate audio and speech codecs. In this paper we describe the components of a novel integrated audio bandwidth extension toolkit (ABET). The ABET toolkit is a combination of two bandwidth extension tools: the Fractal Self-Similarity Model (FSSM) for signal spectrum; and, Accurate Spectral Replacement (ASR). Combination of these two tools, which are applied directly to high frequency resolution representation of the signal such as the Modified Cosine Transform (MDCT), has several benefits for increased accuracy and coding efficiency of the high frequency signal components. At the same time the combination of the two tools entails a number of importation algorithmic and perceptual considerations. In this paper we describe the components of the ABET bandwidth extension toolkit in detail. Algorithmic details, audio demonstrations, and comparison to other audio coding schemes will be presented. Additional information and audio samples are available at http://www.atc-labs.com/abet/.
Presentation is scheduled to begin at 14:00
Convention Paper 6788 (Purchase now)
P20-2 Evaluation of Real-Time Transport Protocol Configurations Using aacPlus—Andreas Schneider, Kurt Krauss, Andreas Ehret, Coding Technologies - Nuremberg, Germany
aacPlus is a highly efficient audio codec that is being used in a growing number of applications where the compressed audio data is encapsulated in a real-time transport protocol and transmitted over error-prone channels. In this paper the implication of packet losses during transmission and techniques to mitigate the impact on the resulting audio quality are discussed. Example transmission channel characteristics are used to show how typical protocol configuration parameters are derived. The benefits of the described techniques are evaluated and verified by setting up a complete simulation chain and performing listening tests.
[Associated Poster Presentation in Session P23, Monday, May 22, at 16:00]
Presentation is scheduled to begin at 14:20
Convention Paper 6789 (Purchase now)
P20-3 Audio Communication Coder—Anibal Ferreira, University of Porto - Porto, Portugal, ATC Labs, Chatham, NJ, USA; Deepen Sinha, ATC Labs - Chatham, NJ, USA
3G mobile and wireless communication networks elicit new ways of multimedia human interaction and communication, notably two-way high-quality audio communication. This is in line with both the consumer expectation of new audio experiences and functionalities, and with the motivation of telecom operators to offer consumers new services and communication modalities. In this paper we describe the design and optimization of a monophonic audio coder (Audio Communication Coder -ACC) that features low-delay coding (< 50 ms) and intrinsic error robustness, while minimizing complexity and achieving competitive coding gains and audio quality at bit rates around 32 kbit/s and higher. ACC source, perceptual, and bandwidth extension tools are described and an emphasis is placed on ACC structural and operational features making it suitable for real-time, two-say audio communication. A few performance results are also presented. Audio demonstrations are available at http://www.atc-labs.com/acc/.
[Associated Poster Presentation in Session P23, Monday, May 22, at 16:00]
Presentation is scheduled to begin at 14:40
Convention Paper 6790 (Purchase now)
P20-4 ISO/IEC MPEG-4 High-Definition Scalable Advanced Audio Coding—Ralf Geiger, Fraunhofer IIS - Erlangen, Germany; Rongshan Yu, Institute for Infocomm Research - Singapore; Jürgen Herre, Fraunhofer IIS - Erlangen, Germany; Susanto Rahardja, Institute for Infocomm Research - Singapore; Sang-Wook Kim, Samsung Electronics - Suwon, Korea; Xiao Lin, Institute for Infocomm Research - Singapore; Markus Schmidt, Fraunhofer IIS - Erlangen, Germany
Recently, the MPEG Audio standardization group has successfully concluded the standardization process on technology for lossless coding of audio signals. This paper provides a summary of the Scalable Lossless Coding (SLS) technology as one of the results of this standardization work. MPEG-4 Scalable Lossless Coding provides a fine-grain scalable lossless extension of the well-known MPEG-4 AAC perceptual audio coder up to fully lossless reconstruction at word lengths and sampling rates typically used for high-resolution audio. The underlying innovative technology is described in detail and its performance is characterized for lossless and near-lossless representation, both in conjunction with an AAC coder and as a stand-alone compression engine. A number of application scenarios for the new technology are also discussed.
[Associated Poster Presentation in Session P23, Monday, May 22, at 16:00]
Presentation is scheduled to begin at 15:00
Convention Paper 6791 (Purchase now)
