High-quality sound coding within 2×64 kbit/s using instantaneous dynamic bit-allocation

Author

Vörös, Paul

Author_Institution

Swiss PTT, Berne, Switzerland

fYear

1988

fDate

11-14 Apr 1988

Firstpage

2536

Abstract

A subband ADPCM (adaptive differential pulse-code modulation) algorithm to encode 15 kHz signals is given. It features dynamic bit allocation based on the log scale factors of the adaptive embedded quantizers and on the roughly quantized instantaneous values of the subband difference signals. Bit allocation is performed for frame lengths equal to one sample interval of the narrowest channel. Side information is transmitted only to maintain octet-sequence integrity and channel synchronization. The receiver can switch dynamically between decoding either 1 or 2×64 kb/s. The algorithm was implemented using three TI 32010 digital signal processors. Results of formal subjective tests are given

Keywords

acoustic signal processing; encoding; pulse-code modulation; 15 kHz; 64 kbit/s; TI 32010 digital signal processors; adaptive differential pulse-code modulation; adaptive embedded quantizers; algorithm; channel synchronization; decoding; dynamic bit allocation; formal subjective tests; frame lengths; log scale factors; octet-sequence integrity; quantized instantaneous values; receiver; side information; sound coding; subband ADPCM; subband difference signals; Acoustic noise; Bit rate; Codecs; Decoding; Delay; Laboratories; Speech; Switches; Testing; Transform coding;

fLanguage

English

Publisher

ieee

Conference_Titel

Acoustics, Speech, and Signal Processing, 1988. ICASSP-88., 1988 International Conference on

Conference_Location

New York, NY

ISSN

1520-6149

Type

conf

DOI

10.1109/ICASSP.1988.197160

Filename

197160