High quality audio communication is a current challenge addressed by the standardisation committees. In this context, ITU and MPEG recently issued standards for high quality coding of both speech and music contents. Transform coding is used and allows quality commensurate with bit rates regardless of the audio content. Up to now, only constant transform sizes were used in these coding schemes since time varying transform needed lookahead for perfect reconstruction, hence adding further delay. In this paper we demonstrate how variable transform sizes can be used without affecting the coding delay. Based on the filterbank theory, a framework avoiding lookahead is presented. The quality improvement offered by the proposed solution is illustrated in the context of MPEG-4 Enhanced Low Delay AAC.
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