EFFICIENT BACKWARD CODING FOR WIDE-BAND SPEECH AND MUSIC SIGNALS (ADPCM-AB).

Shinji Hayashi, Masaaki Honda, Nobuhiko Kitawaki

Research output: Contribution to journalArticle

Abstract

This paper proposes a new backward speech coding system (ADPCM-AB) for wideband (about 8 kHz) speech and music signals. This system uses a split-band adaptive predictive coding scheme with a PARCOR lattice filter and backward dynamic bit allocation over subbands and stereo channels. Linear prediction coefficients and dynamic bit allocation parameters are calculated using locally decoded residual signals. Accordingly, delay time is only 5 milliseconds and transmission of side-information parameters is unnecessary. Paired comparison listening tests confirm that the speech quality of ADPCM-AB is very close to that of forward coding system.

Original languageEnglish
Pages (from-to)363-368
Number of pages6
JournalReports of the Electrical Communication Laboratory
Volume36
Issue number3
Publication statusPublished - 1988 May
Externally publishedYes

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Speech coding
Time delay

ASJC Scopus subject areas

  • Engineering(all)

Cite this

EFFICIENT BACKWARD CODING FOR WIDE-BAND SPEECH AND MUSIC SIGNALS (ADPCM-AB). / Hayashi, Shinji; Honda, Masaaki; Kitawaki, Nobuhiko.

In: Reports of the Electrical Communication Laboratory, Vol. 36, No. 3, 05.1988, p. 363-368.

Research output: Contribution to journalArticle

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