A 7 kHz bandwidth, 32 kbps speech coder for ISDN

Author

Quackenbush, Schuyler

Author_Institution

AT&T Bell Lab., Murray Hill, NJ, USA

fYear

1991

fDate

14-17 Apr 1991

Firstpage

1

Abstract

The author describes a voice codec with a 20 to 7000 Hz bandwidth that compresses to 2 bits per sample at a sample rate of 16 kHz. The algorithm is based on a transform coder that employs dynamic bit allocation in the frequency domain using a perceptual noise weighting criterion, and by Huffman compression of the quantized transform signal. It is well suited to ISDN (integrated services digital network) channels since it permits two coded voice signals to be carried on a single 64 kbps ISDN B channel. A two-channel coder has been implemented in real-time using three AT&T DSP32C signal processors and runs on the public switched ISDN network. The one-way communication delay of this hardware is 75 msec

Keywords

ISDN; encoding; speech analysis and processing; vocoders; 20 Hz to 7 kHz; 32 kbit/s; 75 msec; AT&T DSP32C signal processors; Huffman compression; ISDN B channel; dynamic bit allocation; frequency domain; integrated services digital network; one-way communication delay; perceptual noise weighting criterion; public switched ISDN network; quantized transform signal; transform coder; two coded voice signals; voice codec; Bandwidth; Bit rate; Codecs; Communication switching; Delay; Frequency domain analysis; ISDN; Signal processing; Signal processing algorithms; Speech;

fLanguage

English

Publisher

ieee

Conference_Titel

Acoustics, Speech, and Signal Processing, 1991. ICASSP-91., 1991 International Conference on

Conference_Location

Toronto, Ont.

ISSN

1520-6149

Print_ISBN

0-7803-0003-3

Type

conf

DOI

10.1109/ICASSP.1991.150264

Filename

150264