Speech compression using wavelet transform

F. W. Zaki, H. Hashish, S. Behiry
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引用次数: 1

Abstract

A great deal of mathematical and engineering analysis depends on methods for representing a complex phenomenon in terms of elementary, well-understood phenomena. Recently wavelet theory has provided a new method for decomposing a function or signal. Wavelet transforms of real signals tend to create low-value coefficients at the finer scale. This property makes it possible to devise compression schemes that quantise the fine-scale coefficients more severely than coefficients at other scales. In this paper, two schemes for speech compression based on wavelet transform are introduced. In the first one, short time segments of speech samples are wavelet transformed and quantised using different types of quantisers. In the second scheme, a linear prediction is carried out using the autocorrelation method and tapped delay line transversal filter, then the prediction residual is wavelet transformed, quantised, and transmitted. A good quality speech reproduction is obtained at compression ratios of 50%-90.36%. The system may find its applications in cellular and satellite communications.
基于小波变换的语音压缩
大量的数学和工程分析依赖于用基本的、众所周知的现象来表示复杂现象的方法。近年来,小波理论为分解函数或信号提供了一种新的方法。实际信号的小波变换往往在更小的尺度上产生低值系数。这一特性使得设计压缩方案成为可能,这些压缩方案比其他尺度的系数更严格地量化精细尺度系数。本文介绍了两种基于小波变换的语音压缩方案。在第一种方法中,对语音样本的短时间片段进行小波变换,并使用不同类型的量化器进行量化。第二种方案采用自相关法和抽头延迟线横向滤波器进行线性预测,然后对预测残差进行小波变换、量化和传输。在50% ~ 90.36%的压缩比下,获得了较好的语音再现质量。该系统可用于蜂窝和卫星通信。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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