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Error Robust Low Delay Audio Coding Based on Subband ADPCM

In this paper we present an approach for error robust audio coding at a medium data rate of about 176kbps (mono, 44.1kHz sampling rate). By combining a delay-free Adaptive Differential Pulse Code Modulation (ADPCM) coding-scheme and a numerically optimized low delay filter bank we achieve a very low algorithmic coding delay of only about 0.5ms. The structure of the codec also allows for a high robustness against random single bit errors and even supports error resilience. Implementation structure, results of a listening test and PEAQ (Perceptual Evaluation of Audio Quality) based objective audio quality evaluation as well as tests of random single bit error performance are given. The presented coding-scheme provides a very good audio quality for vocals and speech. For most of the critical signals the audio quality can still be denoted as acceptable. Tests of random single bit error performance show good results for error rates up to 10 to the -4.

Authors: Preihs, Stephan; Ostermann, Jörn
Affiliation: Institute for Information Processing, Leibniz Universität Hannover, Hannover, Germany
AES Convention:131 (October 2011) Paper Number:8532 Import into BibTeX
Subject:Low Bit-Rate Audio Coding

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