ADPCM with a multiquantizer for speech coding

Author

Taniguchi, Tomohiko ; Unagami, Shigeyuki ; Iseda, Kohei ; Tominaga, Syozi

Author_Institution

Fujitsu Labs. Ltd., Kawasaki, Japan

Volume

6

Issue

2

fYear

1988

fDate

2/1/1988 12:00:00 AM

Firstpage

410

Lastpage

424

Abstract

A speech coding algorithm with low complexity and a short processing delay is introduced. The proposed algorithm is ADPCM (adaptive digital pulse code modulation) with a multiquantizer (ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM coders with different characteristics. Then the optimum ADPCM coder with minimum error power is dynamically selected for each frame. A 16-kb/s codec based on this algorithm has been implemented using two general-purpose digital signal processors (MB8764) with 8.3 ms of total processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s; with postfiltering the segmental SNR was increased to 23-25 dB. Combined with the time domain compression scheme, the algorithm can be easily applied to 8-kb/s coding. It is also extensible to variable-rate coding

Keywords

codecs; digital filters; encoding; filtering and prediction theory; pulse-code modulation; speech analysis and processing; vocoders; voice equipment; 16 kbits/s; 23 to 25 dB; 8 kbits/s; ADPCM; adaptive digital pulse code modulation; codec; error power; general-purpose digital signal processors; multiquantizer; postfiltering; processing delay; speech coding algorithm; time domain compression; variable-rate coding; Bit rate; Codecs; Delay; Hardware; Noise level; Performance evaluation; Quantization; Signal to noise ratio; Speech coding; Speech enhancement;

fLanguage

English

Journal_Title

Selected Areas in Communications, IEEE Journal on

Publisher

ieee

ISSN

0733-8716

Type

jour

DOI

10.1109/49.616

Filename

616