Low delay audio compression using predictive coding

Gerald Schuller, Aki Härmä

Research output: Chapter in Book/Report/Conference proceedingConference article in proceedingAcademicpeer-review

Abstract

A low delay audio coding scheme for communications applications is proposed. Its compression ratio is comparable to current state-of-the-art audio coding schemes, but with a much lower delay. The source of delay in conventional audio coding are the filters for the subband coding, and the block switching of the filter bank. The block switching leads to high peaks in bit-rate which necessitates a large bit rate buffer to smooth the bit rate for a transmission channel. To avoid or reduce these delays, we replace the subband coding by predictive coding, and the hard switching of the filter bank by soft switching of the predictors. The overall delay becomes 6 ms at 32 kHz sampling rate. A subjective listening test with bit-rates around 64 kb/s for mono signals shows that the new scheme has a comparable quality to a conventional state-of-the-art coder (PAC).

Original languageEnglish
Title of host publication2002 IEEE International Conference on Acoustics, Speech, and Signal Processing
Pages1853-1856
Number of pages4
Volume2
DOIs
Publication statusPublished - 2002
Externally publishedYes

Publication series

SeriesICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings
ISSN1520-6149

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