Fourth European Conference on Speech Communication and Technology

Madrid, Spain
September 18-21, 1995

Speech Coding Based on the Discrete-Time Wavelet Transform and Human Auditory System Properties

Dan Stefanoiu (1,2), Radwan Kastantin (1), Gang Feng (1)

(1) Institut de la Communication Pailee - URA CNRS 368, Grenoble, France
(2) University "Politehnica" of Bucharest, Department of Automatic Control & Computer Science, Bucharest, Romania

In this paper, we present a new speech coding algorithm with variable bit rate for telephonic domain. The properties of the Discrete-Time Wavelet Transform and several human psycho-acoustic features are integrated in order to obtain a high quality of coded signal, close to original. This approach allows to reduce the mean bit rate in the range of 10 kbit/s. We describe there the main steps of the coding algorithm. More specifically, we provide the solutions to the following specific problems: choosing the type of the Wavelet Transform, selecting the wavelets coefficients to send, quantifying all sending information (coefficients values and auxiliary) and generating the associated codes.

Full Paper

Bibliographic reference.  Stefanoiu, Dan / Kastantin, Radwan / Feng, Gang (1995): "Speech coding based on the discrete-time wavelet transform and human auditory system properties", In EUROSPEECH-1995, 661-664.